Commit Graph

89 Commits

Author SHA1 Message Date
0eb15ed7b8 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}
2015-12-17 11:04:24 +00:00
726b1f7a14 Removed dummy "mediastreamsignaling.h"
Review URL: https://codereview.webrtc.org/1460483005

Cr-Commit-Position: refs/heads/master@{#10717}
2015-11-19 20:01:10 +00:00
521ed7bf02 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}
2015-11-19 03:42:00 +00:00
318166bed7 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}
2015-11-19 03:03:46 +00:00
2764e1027a 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}
2015-11-19 02:02:40 +00:00
c1aeaf0dc3 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}
2015-10-15 14:26:17 +00:00
d59daf8023 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}
2015-10-14 22:02:50 +00:00
ab9b2d1516 Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ )
Reason for reland:
The original CL actually didn't break browser_tests; it was
just a coincidence that it started failing.

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

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

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

Cr-Commit-Position: refs/heads/master@{#10277}
2015-10-14 18:33:20 +00:00
fc648b6d93 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}
2015-10-13 23:43:11 +00:00
97c3929354 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}
2015-10-13 20:23:48 +00:00
0c4e06b4c6 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}
2015-10-07 10:23:32 +00:00
6caafbe5b6 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}
2015-10-05 19:43:33 +00:00
456696a9c1 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}
2015-10-01 04:49:02 +00:00
27dc29b0df 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}
2015-10-01 02:23:15 +00:00
4fe3c9a773 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}
2015-10-01 01:49:17 +00:00
facbbecb51 Remove use of DeviceManager from ChannelManager.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1346153002

Cr-Commit-Position: refs/heads/master@{#10042}
2015-09-24 07:41:59 +00:00
cbecd358e0 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}
2015-09-23 18:50:31 +00:00
a81a42f584 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}
2015-09-23 09:24:27 +00:00
47ee2f3b9f 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}
2015-09-22 22:08:31 +00:00
8902433a43 Revert "TransportController refactoring."
This reverts commit 9af63f473e1d0d6c47a741a046c41642dfc1c178.

Cr-Commit-Position: refs/heads/master@{#9994}
2015-09-18 20:50:31 +00:00
9af63f473e 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}
2015-09-18 19:56:02 +00:00
b071a19019 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}
2015-09-17 14:43:06 +00:00
f3ecdb981c 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}
2015-09-08 10:12:07 +00:00
d82819892a 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}
2015-08-27 08:12:37 +00:00
0c0226408d 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}
2015-08-05 10:26:01 +00:00
be24c94c95 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}
2015-06-22 22:06:50 +00:00
8e6fd46cc3 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}
2015-06-02 07:25:03 +00:00
7fb711f683 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}
2015-04-22 13:30:33 +00:00
7c027b64ae 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}
2015-04-22 11:21:10 +00:00
c04a97f054 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
2015-03-16 19:32:23 +00:00
b01c707209 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
2015-03-13 20:05:46 +00:00
d3900296ae 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
2015-03-12 16:36:15 +00:00
92f4018d80 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
2015-03-04 15:25:44 +00:00
14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: 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
2015-03-04 13:04:54 +00:00
058b1f17ac 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
2015-03-04 08:55:16 +00:00
7bea1ffe77 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
2015-03-04 01:38:49 +00:00
1ed6224eaf 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
2015-02-19 13:57:43 +00:00
8ad05b7628 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
2015-02-19 13:00:46 +00:00
652bc37a07 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
2015-02-18 23:51:22 +00:00
322a564f49 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
2015-02-03 22:10:13 +00:00
4fb7e25843 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
2015-01-21 11:36:18 +00:00
5f93d0a140 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
2015-01-20 21:36:13 +00:00
8e327c45d0 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
2015-01-19 20:41:26 +00:00
43e54e36bf 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
2015-01-19 17:34:23 +00:00
5b76fd79df 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
2015-01-19 16:49:33 +00:00
487a444215 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
2015-01-15 22:55:07 +00:00
61c1247224 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
2015-01-15 06:53:07 +00:00
950c518251 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
2014-12-16 23:01:31 +00:00
55360ae402 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
2014-12-16 05:28:10 +00:00
aaf02cc2d4 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
2014-12-15 23:03:10 +00:00