| 0eb15ed7b806125774bd13fb214aeb403e2c6857 |
|
17-Dec-2015 |
kwiberg <kwiberg@webrtc.org> |
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
We can now use std::move instead!
This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.
Review URL: https://codereview.webrtc.org/1460043002
Cr-Commit-Position: refs/heads/master@{#11064}
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 726b1f7a1467a33b1c3feedff84fca953f7f9c1d |
|
19-Nov-2015 |
perkj <perkj@webrtc.org> |
Removed dummy "mediastreamsignaling.h"
Review URL: https://codereview.webrtc.org/1460483005
Cr-Commit-Position: refs/heads/master@{#10717}
/external/webrtc/talk/app/webrtc/statscollector_unittest.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/app/webrtc/statscollector_unittest.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/app/webrtc/statscollector_unittest.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/app/webrtc/statscollector_unittest.cc
|
| 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/app/webrtc/statscollector_unittest.cc
|
| d59daf8023286d63a1b6c8af82eedb684181c1eb |
|
15-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Merging BaseSession code into WebRtcSession.
After the TransportController CL, BaseSession does little more than
hold a state and an error, and act as an intermediary for the
TransportController. So it doesn't make sense for it to be its own
class.
Review URL: https://codereview.webrtc.org/1397973002
Cr-Commit-Position: refs/heads/master@{#10281}
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| ab9b2d1516cad017c6e0236c468934582530c965 |
|
14-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ )
Reason for reland:
The original CL actually didn't break browser_tests; it was
just a coincidence that it started failing.
Original issue's description:
> Revert of Moving MediaStreamSignaling logic into PeerConnection. (patchset #10 id:180001 of https://codereview.webrtc.org/1393563002/ )
>
> Reason for revert:
> Broke browser_tests on Mac. Still need to investigate the cause.
>
> Original issue's description:
> > Moving MediaStreamSignaling logic into PeerConnection.
> >
> > This needs to happen because in the future, m-lines will be offered
> > based on the set of RtpSenders/RtpReceivers, rather than the set of
> > tracks that MediaStreamSignaling knows about.
> >
> > Besides that, MediaStreamSignaling was a "glue class" without
> > a clearly defined role, so it going away is good for other
> > reasons as well.
> >
> > Committed: https://crrev.com/97c392935411398b506861601c82e31d95c591f0
> > Cr-Commit-Position: refs/heads/master@{#10268}
>
> TBR=pthatcher@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/fc648b6d934e936f4d9a32c813364b331536ec3b
> Cr-Commit-Position: refs/heads/master@{#10269}
TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1404473005
Cr-Commit-Position: refs/heads/master@{#10277}
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| fc648b6d934e936f4d9a32c813364b331536ec3b |
|
14-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Revert of Moving MediaStreamSignaling logic into PeerConnection. (patchset #10 id:180001 of https://codereview.webrtc.org/1393563002/ )
Reason for revert:
Broke browser_tests on Mac. Still need to investigate the cause.
Original issue's description:
> Moving MediaStreamSignaling logic into PeerConnection.
>
> This needs to happen because in the future, m-lines will be offered
> based on the set of RtpSenders/RtpReceivers, rather than the set of
> tracks that MediaStreamSignaling knows about.
>
> Besides that, MediaStreamSignaling was a "glue class" without
> a clearly defined role, so it going away is good for other
> reasons as well.
>
> Committed: https://crrev.com/97c392935411398b506861601c82e31d95c591f0
> Cr-Commit-Position: refs/heads/master@{#10268}
TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1403633005
Cr-Commit-Position: refs/heads/master@{#10269}
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 97c392935411398b506861601c82e31d95c591f0 |
|
13-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Moving MediaStreamSignaling logic into PeerConnection.
This needs to happen because in the future, m-lines will be offered
based on the set of RtpSenders/RtpReceivers, rather than the set of
tracks that MediaStreamSignaling knows about.
Besides that, MediaStreamSignaling was a "glue class" without
a clearly defined role, so it going away is good for other
reasons as well.
Review URL: https://codereview.webrtc.org/1393563002
Cr-Commit-Position: refs/heads/master@{#10268}
/external/webrtc/talk/app/webrtc/statscollector_unittest.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/statscollector_unittest.cc
|
| 6caafbe5b6b777b309a6eb90a02cf54d5106fb9b |
|
05-Oct-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Convert uint16_t to int for WebRTC cipher/crypto suite.
This is a follow up CL on https://codereview.webrtc.org/1337673002
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1377733004 .
Cr-Commit-Position: refs/heads/master@{#10175}
/external/webrtc/talk/app/webrtc/statscollector_unittest.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/app/webrtc/statscollector_unittest.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/app/webrtc/statscollector_unittest.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/app/webrtc/statscollector_unittest.cc
|
| facbbecb516547adc2ac684c8e0be95ad79dfd88 |
|
24-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove use of DeviceManager from ChannelManager.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1346153002
Cr-Commit-Position: refs/heads/master@{#10042}
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 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/app/webrtc/statscollector_unittest.cc
|
| 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/app/webrtc/statscollector_unittest.cc
|
| 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/app/webrtc/statscollector_unittest.cc
|
| 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/app/webrtc/statscollector_unittest.cc
|
| 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/app/webrtc/statscollector_unittest.cc
|
| b071a19019a0a2173cc139c960d6ef6946a1c581 |
|
17-Sep-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Full use of NnChannel::SetSendParameters and NnChannel::SetRecvParameters.
SetOptions(), SetMaxBandwidth(), Set[Send|Recv]RtpHeaderExtensions(), Set[Send|Recv]Codecs() are now private.
BUG=webrtc:4690
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1327933002 .
Cr-Commit-Position: refs/heads/master@{#9973}
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| f3ecdb981c172cdfafbe92c939eb25ddcc1ae96e |
|
08-Sep-2015 |
Henrik Boström <hbos@webrtc.org> |
Replacing SSLIdentity* with scoped_refptr<RTCCertificate> in TransportChannel layer.
BUG=webrtc:4927
R=tommi@webrtc.org, torbjorng@webrtc.org
Review URL: https://codereview.webrtc.org/1304043008 .
Cr-Commit-Position: refs/heads/master@{#9885}
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| d82819892a382899a82ced756a9922a84ca9ca98 |
|
27-Aug-2015 |
Henrik Boström <hbos@webrtc.org> |
Replaces SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::Transport layer.
Why the replacements? Mainly two reasons:
1) RTCCertificate owns the identity and as long as things are referencing the identity there should be a scoped_refptr reference to the RTCCertificate. Handing out raw pointers is less memory safe.
2) With the latest RFC, an RTCCertificate should be sufficient for specifying a crypto cert and the code should be updated to use RTCCertificate instead of SSLIdentity directly.
This replace work is split up into multiple CLs. In this CL...
- WebRtcSessionDescriptionFactory is updated to use RTCCertificate over SSLIdentity.
- WebRtcSessionDescriptionFactory::SignalCertificateReady is connected to WebRtcSession::OnCertificateReady and WebRtcSession is updated to use RTCCertificate.
- The cricket::Transport and related classes are updated to use RTCCertificate. These are called from WebRtcSession::OnCertificateReady.
BUG=webrtc:4927
R=tommi@webrtc.org, torbjorng@webrtc.org
Review URL: https://codereview.webrtc.org/1312643004 .
Cr-Commit-Position: refs/heads/master@{#9794}
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 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/app/webrtc/statscollector_unittest.cc
|
| be24c94c95056e4f0a22039f25f2fa8a27be6b66 |
|
23-Jun-2015 |
jbauch <jbauch@webrtc.org> |
Set / verify stats report timestamps.
This CL updates the track report timestamps which were fixed at "0" before
and updates the timestamps in reports for local audio tracks.
Also the timestamps are checked in various tests to make sure no "0" is
returned.
Original CL is at https://webrtc-codereview.appspot.com/51829004/
BUG=webrtc:4316
TBR=hta@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1204493002
Cr-Commit-Position: refs/heads/master@{#9485}
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 8e6fd46cc324f8946e68396edcb252ffaf2d4579 |
|
02-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Route time-stretching metrics through libjingle
This change connects currentAccelerateRate and currentPreemptiveRate
in webrtc::NetworkStatistics, through corresponding variables in
VoiceReceiverInfo, to googAccelerateRate and googPreemptiveExpandRate.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50179004
Cr-Commit-Position: refs/heads/master@{#9350}
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 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/app/webrtc/statscollector_unittest.cc
|
| 7c027b64ae53a29bc528b4241cc540694c239304 |
|
22-Apr-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Enable more Clang warnings for talk/
BUG=4242
R=andresp@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46999004
Cr-Commit-Position: refs/heads/master@{#9053}
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| c04a97f054348909c5b0c24369fb4272c2c16041 |
|
16-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Move from BaseSession::GetStats to WebRtcSession::GetTransportStats
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
Review URL: https://webrtc-codereview.appspot.com/45639004
Cr-Commit-Position: refs/heads/master@{#8739}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8739 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| b01c707209eff893223ed7af1e5fdb75b34a22a4 |
|
13-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Use a NULL session in unit tests that don't actually use the session.
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/49379004
Cr-Commit-Position: refs/heads/master@{#8721}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8721 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| d3900296ae4416de2ea21be4548ea4adba8f3280 |
|
12-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Use a variant for storing stats values in StatsCollector code.
This cuts down on the amount of string copying we currently do and paves the way for separating the code that fetches the stats from the code that populates the stats reports. As is, that code is intertwined, so we populate the stats on both signaling and worker thread.
I'm also adding some documentation and TODOs for further improvements.
BUG=2822
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47459004
Cr-Commit-Position: refs/heads/master@{#8700}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8700 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 92f4018d80ec8b092b7c1a35528e57e926f75111 |
|
04-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Start using std::map for Values in the statscollector. This is in preparaton for more work which will cut down on the string copying work we do.
Rename "AddValue" methods to AddXxx where Xxx is the type being added. Moving forward, we'll support those types natively without conversion to string.
Normalizing the extraction code to have fewer places that add the same stats and data driven additions to reports instead of multiple call sites.
BUG=2822
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47369004
Cr-Commit-Position: refs/heads/master@{#8597}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8597 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 14665ff7d4024d07e58622f498b23fd980001871 |
|
04-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: https://chromium.googlesource.com/chromium/src/+/e144d30..6fdb142/tools/clang/scripts/update.sh
Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h
The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"`
which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override
Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h
Remaining uses of OVERRIDE was fixed by search+replace.
Manual edits were done to fix virtual destructors that were
overriding inherited ones.
Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc
This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.
BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41069004
Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 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/app/webrtc/statscollector_unittest.cc
|
| 7bea1ffe772e837d96f8faa5c9dd06e531b95379 |
|
04-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Expose negotiated ciphers through stats API.
Use the new internal API to expose the negotiated SRTP/SSL ciphers
through the stats API.
This is a follow-up to https://webrtc-codereview.appspot.com/37209004.
BUG=3976
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35169004
Cr-Commit-Position: refs/heads/master@{#8584}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8584 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 1ed6224eafc7816f25d1906e4d709afdf2ad8f0f |
|
19-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Revert r8430 "Remove dead stats from Video{Sender,Receiver}Info."
This breaks compilation outside this codebase that needs to have it
removed before.
BUG=4322
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42009004
Cr-Commit-Position: refs/heads/master@{#8432}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8432 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 8ad05b76281e73f92051125aee81d85227c6a9bc |
|
19-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove dead stats from Video{Sender,Receiver}Info.
These stats are neither filled nor plumbed further and might as well be
removed (as proven by how easy they were to remove).
BUG=
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39219004
Cr-Commit-Position: refs/heads/master@{#8430}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8430 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 652bc37a07f5ab2559fd217c22be391b45af5b53 |
|
19-Feb-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Adding two new stats to StatsReport.
A follow up of r8415. This is to post the data to the StatsReport.
BUG=3867
TEST=chromium + netem + apprtc + chrome://webrtc-internals
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38139004
Cr-Commit-Position: refs/heads/master@{#8423}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8423 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 322a564f49d9c995cfffbaabd3d8c5d5aa326e86 |
|
03-Feb-2015 |
decurtis@webrtc.org <decurtis@webrtc.org> |
Fix datachannel stats id and timestamp.
Makes the id now be "datachannel_#####" where '####' is the id number for the datachannel.
Adds a timestamp to the data channel reports.
Implements unit tests to verify that the timestamp is set correctly.
BUG=1805
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33119004
Cr-Commit-Position: refs/heads/master@{#8236}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8236 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 4fb7e2584326050a707aef544028fa9cb616ec89 |
|
21-Jan-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Update StatsReport and by extension StatsCollector to reduce data copying.
Summary of changes:
* We're now using an enum for types instead of strings which both eliminates unecessary string creations+copies and further restricts the type to a known set at compile time.
* IDs are now a separate type instead of a string, copying of Values is not possible and values are const to allow grabbing references outside of the statscollector.
* StatsReport member variables are no longer public.
* Consolidated code in StatsCollector (e.g. merged PrepareLocalReport and PrepareRemoteReport).
* Refactored methods that forced copies of string (e.g. ExtractValueFromReport).
* More asserts for thread correctness.
* Using std::list for the StatsSet instead of a set since order is not important and updates are more efficient in list<>.
BUG=2822
R=hta@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8110 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.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/statscollector_unittest.cc
|
| 8e327c45d0940fd5bc46c3fe8d24363be07706ac |
|
19-Jan-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Update StatsCollector's interface in preparation of more changes.
This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.
The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.
The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.
BUG=2822
R=perkj@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8095
Review URL: https://webrtc-codereview.appspot.com/36829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8097 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 43e54e36bff3f6159e9c7ac0aa40beafca485c56 |
|
19-Jan-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Revert 8095 "Update StatsCollector's interface in preparation of..."
> Update StatsCollector's interface in preparation of more changes.
>
> This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.
>
> The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.
>
> The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.
>
> BUG=2822
> R=perkj@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/36829004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8096 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 5b76fd79dfbfa78f1b034c0698771298cd15f175 |
|
19-Jan-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Update StatsCollector's interface in preparation of more changes.
This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.
The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.
The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.
BUG=2822
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8095 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 487a4442152e2c70146aa2d2c6ccb370233c056c |
|
15-Jan-2015 |
decurtis@webrtc.org <decurtis@webrtc.org> |
Add stats collection for the data channel.
BUG=1805
R=bemasc@chromium.org, hta@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8083 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.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/statscollector_unittest.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/statscollector_unittest.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/statscollector_unittest.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/statscollector_unittest.cc
|
| 69bc5a300fe27448bcb61670f2800d3919ed2975 |
|
15-Dec-2014 |
tommi@webrtc.org <tommi@webrtc.org> |
Add thread asserts to StatsCollector.
Also adding a "ForTest" postfix to a test-only method.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7894 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.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/statscollector_unittest.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/statscollector_unittest.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/statscollector_unittest.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/statscollector_unittest.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/statscollector_unittest.cc
|
| 1ecbe45c7e4c9142896cb2810d699558518f4f28 |
|
14-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 77689511-> 77696841
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7449 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 5b06b06cc0ef5a051fa5b1ed687218a21639d93e |
|
15-Aug-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6897 (i.e. Reland 6863) - "Revert 6863 "Refactor StatsCollector and associated..."
The bot that had the problem was using an old version of STL, so relanding.
> Revert 6863 "Refactor StatsCollector and associated types."
>
> Breaks chrome compilation on Mac:
>
> /Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/vector.tcc:252:8:
> error: no matching constructor for initialization of
> 'webrtc::StatsReport'
> _Tp __x_copy = __x;
> ^ ~~~
> /Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/stl_vector.h:608:4:
> note: in instantiation of member function
> 'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
> >::_M_insert_aux' requested here
> _M_insert_aux(end(), __x);
> ^
> ../../content/renderer/media/mock_peer_connection_impl.cc:282:11:
> note: in instantiation of member function
> 'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
> >::push_back' requested here
> reports.push_back(report1);
> ^
> ../../third_party/libjingle/source/talk/app/webrtc/statstypes.h:49:3:
> note: candidate constructor not viable: requires 0 arguments, but 1
> was provided
> StatsReport() : timestamp(0) {}
>
>
>
> > Refactor StatsCollector and associated types.
> > * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> > * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> > * Report ids are now const.
> > * Copying of data has been greatly reduced.
> > * This change includes preparation work for making GetStats fully async.
> >
> > This is a reland of r6778 which was reverted due to fyi bots failing.
> > I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.
> >
> > R=xians@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/15119004
>
> TBR=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/21169004
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6908 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 22fa032f223e6b6210d569c8ae813c1a1a6edc07 |
|
14-Aug-2014 |
niklas.enbom@webrtc.org <niklas.enbom@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6863 "Refactor StatsCollector and associated types."
Breaks chrome compilation on Mac:
/Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/vector.tcc:252:8:
error: no matching constructor for initialization of
'webrtc::StatsReport'
_Tp __x_copy = __x;
^ ~~~
/Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/stl_vector.h:608:4:
note: in instantiation of member function
'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
>::_M_insert_aux' requested here
_M_insert_aux(end(), __x);
^
../../content/renderer/media/mock_peer_connection_impl.cc:282:11:
note: in instantiation of member function
'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
>::push_back' requested here
reports.push_back(report1);
^
../../third_party/libjingle/source/talk/app/webrtc/statstypes.h:49:3:
note: candidate constructor not viable: requires 0 arguments, but 1
was provided
StatsReport() : timestamp(0) {}
> Refactor StatsCollector and associated types.
> * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> * Report ids are now const.
> * Copying of data has been greatly reduced.
> * This change includes preparation work for making GetStats fully async.
>
> This is a reland of r6778 which was reverted due to fyi bots failing.
> I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.
>
> R=xians@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/15119004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6897 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.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/statscollector_unittest.cc
|
| 730bf30da75514f22fc1869a93e130e582a8e045 |
|
11-Aug-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.
This is a reland of r6778 which was reverted due to fyi bots failing.
I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6863 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.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/statscollector_unittest.cc
|
| 185636cf708890c04d540be1cf4be4a867c984c0 |
|
25-Jul-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert of 6778 "Refactor StatsCollector and associated types."
Breakes FYI bots.
BUG=N/A
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6783 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 190d269c0f3a1857a11bb12d61c758361737b70a |
|
25-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.
(This is a reland of the original attempt in r6747)
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6778 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 06b04ec4ab5f0366fa20b286588c63f74141ea11 |
|
24-Jul-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix a crash in statscollector.cc caused by invoking methods on the worker thread which destroys the Transport.
BUG=3579
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6776 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 39f831fbb08992bf7abd0fd05d2af0fc1f8756d0 |
|
24-Jul-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Re-revert of 6747 "Refactor StatsCollector and associated types."
Breakes FYI bots.
BUG=N/A
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6772 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 437d57db5b3c7545d5edba2f123f8bcbdf2f80a3 |
|
23-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71775619-> 71778545
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6771 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 8c7e3291a956901f00aab813867cc21a35709d4d |
|
23-Jul-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6747 "Refactor StatsCollector and associated types."
Breakes FYI bots.
BUG=N/A
TBR=ajm@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6770 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 756b8462ebcd9d74234384239428d05f64907fa2 |
|
21-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.
R=xians@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=6745
Review URL: https://webrtc-codereview.appspot.com/18819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6747 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| fd61a1d693840b3e177cade683f3e6d3d0119a9d |
|
21-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6745 "Refactor StatsCollector and associated types."
Broke build on android.
> Refactor StatsCollector and associated types.
> * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> * Report ids are now const.
> * Copying of data has been greatly reduced.
> * This change includes preparation work for making GetStats fully async.
>
> R=xians@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/18819004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6746 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 647e05cfcdb5028a5556ec0268a65ea6794f47a8 |
|
21-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6745 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 242068d58cc01640aa9f733fa67f078fc65c4ae5 |
|
14-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
A step towards changing StatsReport::Value::name to an enum.
The stats reporting code does a lot of unnecessary string copying.
This is a step in the direction of removing that and forcing use of only known constants.
This is a reland of an already reviewed cl that got reverted by mistake.
TBR=xians@google.com
Review URL: https://webrtc-codereview.appspot.com/12989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6678 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 03505bcb7a369add7abfe306004e7803ab096f21 |
|
14-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make StatsCollector depend on always having a valid session pointer.
This is required since the session pointer is currently used on multiple threads but there's no synchronization code to guard it.
I'm removing the set_session() method and session() getter since they would cause problems if used without synchronization.
This is a reland of an already reviewed cl that got reverted by mistake.
TBR=xians@google.com
Review URL: https://webrtc-codereview.appspot.com/13959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6677 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 01bda2068bebb65a610c0d951f938db5dd028394 |
|
09-Jul-2014 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixed the stats problem when new track is using the same ssrc as the previous track.
Before this patch, when switching from voice mode to stereo mode, the stats won't be updated because StatsCollector binded the ssrc report with the old track, so the report can't be updated by the new track.
This patch fixes the porblem by changing the ssrc report track id to use the new track id.
TEST=libjingle_peerconnection_unittest --gtest_filter="*StatsCollectorTest*"
R=hta@chromium.org
Review URL: https://webrtc-codereview.appspot.com/17859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6632 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 7e71b77f8aab5b7a6f2b669c16f90ec9a4b4609c |
|
13-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69102234-> 69116997
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6424 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 4cb012858f7461015e405c0c2cfc4b9f10a086ce |
|
12-Jun-2014 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixed GetStats when local and remote track are using the same ssrc.
R=hta@chromium.org, kjellander@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6414 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| c14f521b1b47f33959bf589ac2937af49db74782 |
|
13-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66887616-> 66900106
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6130 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 3e01e0b16cbde481241b9bcfdbbdd591cd920b99 |
|
13-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66867790-> 66887616
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6128 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| ffe2620c97c2b7bfe42b04453b5a981dbf1e5f06 |
|
19-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63352036-> 63363208
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5731 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| b90991dade9139e5c14c3b616a9eff07b9d6fdda |
|
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/app/webrtc/statscollector_unittest.cc
|
| 40b3b68cdf47d7c9c3b57fca5d0a372292025f9e |
|
03-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle 62364298->62472237
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5632 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| 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/app/webrtc/statscollector_unittest.cc
|
| 0de29504ab7ac923401c8e4e154f3b72038dbcc2 |
|
13-Feb-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5545 "Update libjingle to 61514460"
> Update libjingle to 61514460
>
> TBR=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/8649004
TBR=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5547 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.cc
|
| e749c9ebdb2eb2a519c72c827e70107cbc56d270 |
|
13-Feb-2014 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 61514460
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5545 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.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/statscollector_unittest.cc
|
| 97077a3ab27259164eb121034b6e0ebe9ba592df |
|
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/app/webrtc/statscollector_unittest.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/statscollector_unittest.cc
|
| 4551b793dea4b5451cbfa13b206b6d11a25081d0 |
|
09-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 53920541.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2371004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4945 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/statscollector_unittest.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/statscollector_unittest.cc
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