Commit Graph

13158 Commits

Author SHA1 Message Date
2f69ce9498 Cleaned out candidateSet member from TMMBRHelp class
leaving that class memberless.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2234783002
Cr-Commit-Position: refs/heads/master@{#13776}
2016-08-16 10:21:44 +00:00
1c814e7b72 iOS: Update MB and JSON configs + enable Goma
Turns out that if mb_type is missing in the JSON, GYP is run the
traditional way instead of having the MB configuration decide.
This turns on MB for those builders.
See https://codereview.chromium.org/2194703002 for how Chromium
switched from GYP->GN.

The JSON environment for GYP and GN is only used during runhooks
step since there are scripts that key on some of these environment variables.
The actual build that is compiled is defined by the MB config, which
is now updated to have component=static_library everywhere for iOS.
With this CL, all configs gets a full GYP+GN environment.

When flipping bots over to GN, the following line will need to be added
in addition to changing mb_type:
"additional_compile_targets": [ "all" ],

Goma was also enabled for all builders to reduce compile time.

BUG=589510
NOTRY=True

Review-Url: https://codereview.webrtc.org/2239643002
Cr-Commit-Position: refs/heads/master@{#13775}
2016-08-16 09:42:06 +00:00
8eb37a39e7 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
Reason for revert:
Failed on Win 10 Chrome FYI.

https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win10%20Tester/builds/3847/steps/content_browsertests/logs/stdio

#
# Fatal error in e:\b\c\b\win_builder\src\third_party\webrtc\base\task_queue_win.cc, line 138
# last system error: 87
# Check failed: ((DWORD)0xFFFFFFFF) != result (4294967295 vs. 4294967295)
#

WebRtcBrowserTest

#

Original issue's description:
> - Add task queue to Call with the intent of replacing the use of one of the process threads.
>
> - Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.
>
> - BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.
>
> - VideoEncoderConfig and VideoSendStream::Config support move semantics.
>
> - The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.
>
> BUG=webrtc:5687
>
> Committed: https://crrev.com/cc168360f41322332860cb075edeb1cde21aa473
> Cr-Commit-Position: refs/heads/master@{#13767}

TBR=tommi@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org,sprang@webrtc.org,pbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/2248713003
Cr-Commit-Position: refs/heads/master@{#13774}
2016-08-16 09:40:59 +00:00
6910537458 Add gn target for audio_device_tests.
Note (for myself) that this depends on https://codereview.webrtc.org/2219653004/ and https://codereview.webrtc.org/2214003002/, it should not be landed before them.

NOTRY=True
BUG=webrtc:6170,webrtc:5949

Review-Url: https://codereview.webrtc.org/2216423002
Cr-Commit-Position: refs/heads/master@{#13773}
2016-08-16 09:17:48 +00:00
70f866c647 Added new mixer to |check_targets| in .gn and fixed include/depend errors.
Also fixed one small chromium-style error in the new mixer.

NOTRY=True

Committed: https://crrev.com/d700bef583d29ba2834ae57b3af7e8d3b8306cb9
Review-Url: https://codereview.webrtc.org/2234293002
Cr-Original-Commit-Position: refs/heads/master@{#13752}
Cr-Commit-Position: refs/heads/master@{#13772}
2016-08-16 09:15:55 +00:00
7522a28051 Removed old probe cluster logic and logic related to ssrcs from DelayBasedBwe.
-Removed the old probe cluster logic and use the new ProbeBitrateEstimator
 instead.
-Removed all logic related to ssrcs from DelayBasedBwe as they have no function
 on the sender side.

BUG=webrtc:5859
R=stefan@webrtc.org, terelius@webrtc.org

Review URL: https://codereview.webrtc.org/2234363002 .

Cr-Commit-Position: refs/heads/master@{#13771}
2016-08-16 08:59:48 +00:00
b7186d0aa7 Migrated GN target :isac_fix_test
Migrated GN target :isac_fix_test from
webrtc/modules/audio_coding/codecs/isac/isacfix_test.gypi

NOTRY=True

BUG=webrtc:6191

Review-Url: https://codereview.webrtc.org/2247233002
Cr-Commit-Position: refs/heads/master@{#13770}
2016-08-16 08:47:26 +00:00
b24b1ceb48 Moving mock classes around so that they may be reused in other unittests
New files, classes moved from statscollector_unittest.cc:
+webrtc/api/test/mock_peerconnection.h
 for MockPeerConnectionFactory and MockPeerConnection
+webrtc/api/test/mock_webrtcsession.h
 for MockWebRtcSession
+webrtc/media/base/test/mock_mediachannel.h
 for MockVideoMediaChannel and MockVoiceMediaChannel

The webrtc/media/base/test folder is new.

BUG=chromium:627816

Review-Url: https://codereview.webrtc.org/2238933002
Cr-Commit-Position: refs/heads/master@{#13769}
2016-08-16 08:19:48 +00:00
88e31a3fd8 Fix warnings, simplify ADM.
This is in preparation for adding a gn target for audio_device_tests.

BUG=webrtc:6170,webrtc:163
NOTRY=True

Review-Url: https://codereview.webrtc.org/2222563002
Cr-Commit-Position: refs/heads/master@{#13768}
2016-08-16 07:56:14 +00:00
cc168360f4 - Add task queue to Call with the intent of replacing the use of one of the process threads.
- Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.

- BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.

- VideoEncoderConfig and VideoSendStream::Config support move semantics.

- The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.

BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/2060403002
Cr-Commit-Position: refs/heads/master@{#13767}
2016-08-16 07:38:51 +00:00
82dda1af5e [WebRTC] Disable DirectX capturer tests if the system does not support it.
A recent DrMemory failure has been detected after change 2099123002. After some
investigation, an uninitialized read has been detected in

NtUserGetThreadDesktop
webrtc::Desktop::GetThreadDesktop
webrtc::ScopedThreadDesktop::ScopedThreadDesktop
webrtc::ScreenCapturerWinGdi::ScreenCapturerWinGdi
webrtc::ScreenCapturer::Create
webrtc::ScreenCapturerTest_UseDirectxCapturer_Test::TestBody

So there are two issues,
1. The Directx capturer won't be triggered as the system does not support it. So
these tests should be disabled in this scenario.
2. An uninitialized read in NtUserGetThreadDesktop -> ScopedThreadDesktop
stacks, which should be suppressed. By default, these suppressions should be
placed in chromium/external with other suppressions.

So this change is a quick fix to the failure, do not use ScreenCapturerWinGdi in
ScreenCaputrerWinDirectx tests.

BUG=

Review-Url: https://codereview.webrtc.org/2247943002
Cr-Commit-Position: refs/heads/master@{#13766}
2016-08-16 02:53:40 +00:00
e1b4d243e2 Skip AUD while extracting SPS and PPS on iOS.
H.264 frames may have AUD before SPS. This change detects AUD and try to extract SPS and PPS behind AUD.

BUG=webrtc:6173

Review-Url: https://codereview.webrtc.org/2209143002
Cr-Commit-Position: refs/heads/master@{#13765}
2016-08-16 01:56:23 +00:00
6c687e72a0 Make prior H264 QP adjustments iOS specific.
BUG=

Review-Url: https://codereview.webrtc.org/2248883002
Cr-Commit-Position: refs/heads/master@{#13764}
2016-08-15 23:38:02 +00:00
3473288296 Remove VERBOSE logs in (android) audio device code.
When playing out, for example, you'd see 3 lines for every call to
PlayoutDelay, which happens quite often (every sample?).

The ones around the Playout/Recording Warning/Error are only once a
second, but they don't seem to add anything. Same with
Process/TimeUntilNextProcess, which just log that the method is called.

BUG=

Review-Url: https://codereview.webrtc.org/2202243004
Cr-Commit-Position: refs/heads/master@{#13763}
2016-08-15 20:41:28 +00:00
43ba317c75 Roll chromium_revision 4b42aa218b..2b53ee0889 (411951:411979)
Change log: 4b42aa218b..2b53ee0889
Full diff: 4b42aa218b..2b53ee0889

No dependencies changed.
No update to Clang.

TBR=

Review-Url: https://codereview.webrtc.org/2242193003
Cr-Commit-Position: refs/heads/master@{#13762}
2016-08-15 19:25:31 +00:00
b1e66110e3 GN: Fix audio_decoder_unittests for android.
Tested on a physical device.

BUG=webrtc:6036
NOTRY=True

Review-Url: https://codereview.webrtc.org/2241293002
Cr-Commit-Position: refs/heads/master@{#13761}
2016-08-15 19:02:08 +00:00
4a1ec1e639 Added ProbeBitrate(bitrate_bps, num_probes) to BitrateProber.
Also some minor cleanup.

BUG=webrtc:5859

Review-Url: https://codereview.webrtc.org/2243823002
Cr-Commit-Position: refs/heads/master@{#13760}
2016-08-15 18:51:12 +00:00
1aee0b5bd9 Remove old methods in AudioTransport, make it pass a gn build
when building with default warnings.

This is in preparation for making a gn target for audio_device_tests.

BUG=webrtc:6170, webrtc:163
NOTRY=True

Review-Url: https://codereview.webrtc.org/2219653004
Cr-Commit-Position: refs/heads/master@{#13759}
2016-08-15 18:46:28 +00:00
c8c71f484e Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #6 id:100001 of https://codereview.webrtc.org/2240163002/ )
Reason for revert:
Breaks downstream code, so revert again. Yay.

Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> (This is a re-land of https://codereview.webrtc.org/2037623002, which
> had to be reverted.)
>
> NOPRESUBMIT=True
>
> Committed: https://crrev.com/dc65ea29b3270ad418050658ad962ddd33ee70c1
> Cr-Commit-Position: refs/heads/master@{#13757}

TBR=perkj@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2245153002
Cr-Commit-Position: refs/heads/master@{#13758}
2016-08-15 18:43:56 +00:00
dc65ea29b3 Move FilePlayer and FileRecorder to Voice Engine
Because Voice Engine was the only user.

(This is a re-land of https://codereview.webrtc.org/2037623002, which
had to be reverted.)

NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2240163002
Cr-Commit-Position: refs/heads/master@{#13757}
2016-08-15 17:36:38 +00:00
f96c51ac05 GN: Add video_capture_tests for Mac
BUG=webrtc:6042
NOTRY=True
TBR=mflodman@webrtc.org

Review-Url: https://codereview.webrtc.org/2245083002
Cr-Commit-Position: refs/heads/master@{#13756}
2016-08-15 16:21:36 +00:00
2a7580124b Revert of CQ: Temporarily disable iOS Simulator trybots (patchset #1 id:1 of https://codereview.webrtc.org/2244183002/ )
Reason for revert:
The try server has been reconfigured to not use remote_run for the webrtc/ios recipe now, and builds are passing.

Original issue's description:
> CQ: Temporarily disable iOS Simulator trybots
>
> BUG=637666
> TBR=ehmaldonado@webrtc.org
> NOTRY=True
>
> Committed: https://crrev.com/414eb181d26a794e17f8e0206fa4a7efc116f75a
> Cr-Commit-Position: refs/heads/master@{#13738}

TBR=ehmaldonado@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=637666

Review-Url: https://codereview.webrtc.org/2242173002
Cr-Commit-Position: refs/heads/master@{#13755}
2016-08-15 16:19:45 +00:00
e34c19c9d7 Clarify some function names in visualization tool.
Review-Url: https://codereview.webrtc.org/2230153003
Cr-Commit-Position: refs/heads/master@{#13754}
2016-08-15 15:47:21 +00:00
2ab1da7c37 Revert of Added new mixer to |check_targets| in .gn and fixed include/depend errors. (patchset #1 id:1 of https://codereview.webrtc.org/2234293002/ )
Reason for revert:
Breaks Chromium FYI builds, for example https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/9219/steps/compile/logs/stdio

Original issue's description:
> Added new mixer to |check_targets| in .gn and fixed include/depend errors.
>
> Also fixed one small chromium-style error in the new mixer.
>
> NOTRY=True
>
> Committed: https://crrev.com/d700bef583d29ba2834ae57b3af7e8d3b8306cb9
> Cr-Commit-Position: refs/heads/master@{#13752}

TBR=kjellander@webrtc.org,aleloi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2241243002
Cr-Commit-Position: refs/heads/master@{#13753}
2016-08-15 14:36:22 +00:00
d700bef583 Added new mixer to |check_targets| in .gn and fixed include/depend errors.
Also fixed one small chromium-style error in the new mixer.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2234293002
Cr-Commit-Position: refs/heads/master@{#13752}
2016-08-15 14:24:13 +00:00
963be23e62 RtpRtcp: Remove the SetSendREDPayloadType and SendREDPayloadType methods
The last in-tree call site recently disappeared, so they were unused.

BUG=webrtc:5922

Review-Url: https://codereview.webrtc.org/2066473002
Cr-Commit-Position: refs/heads/master@{#13751}
2016-08-15 14:08:39 +00:00
8f956dead6 FakeTiming added, an implementation of Timing that can be used for
tests.

Note: The webrtc/base/test/ folder is new.

Currently not used, I intend to use this in another CL.

BUG=chromium:627816
NOPRESUBMIT=TRUE
NOTRY=TRUE

Review-Url: https://codereview.webrtc.org/2238073003
Cr-Commit-Position: refs/heads/master@{#13750}
2016-08-15 14:00:06 +00:00
96dbc8f4b4 Adding comment regarding the disabling the flaky test VolumeTest.ManualRequiresMicrophoneCanSetMicrophoneVolumeWithAgcOff
NOTRY=True
TBR=henrika@webrtc.org
BUG=webrtc:6206

Review-Url: https://codereview.webrtc.org/2247733002
Cr-Commit-Position: refs/heads/master@{#13749}
2016-08-15 13:38:56 +00:00
3ab6614d10 Add video_loopback to gn.
BUG=webrtc:5949
NOTRY=True

Review-Url: https://codereview.webrtc.org/2236473002
Cr-Commit-Position: refs/heads/master@{#13748}
2016-08-15 13:29:19 +00:00
92c09509bd Make CameraCapturer.switchCamera try again if session is still opening.
R=magjed@webrtc.org

Review-Url: https://codereview.webrtc.org/2238263002
Cr-Commit-Position: refs/heads/master@{#13747}
2016-08-15 13:19:40 +00:00
d7d05f8056 Disabling the test VolumeTest.ManualRequiresMicrophoneCanSetMicrophoneVolumeWithAgcOff
as it has been found to be flaky.

NOTRY=True
TBR=henrika@webrtc.org
BUG=webrtc:6206

Review-Url: https://codereview.webrtc.org/2248633003
Cr-Commit-Position: refs/heads/master@{#13746}
2016-08-15 13:13:21 +00:00
da07af2441 Roll chromium_revision 7f405ec2b6..4b42aa218b (411933:411951)
Change log: 7f405ec2b6..4b42aa218b
Full diff: 7f405ec2b6..4b42aa218b

No dependencies changed.
No update to Clang.

TBR=

Review-Url: https://codereview.webrtc.org/2247703002
Cr-Commit-Position: refs/heads/master@{#13745}
2016-08-15 12:29:00 +00:00
3e3ebe6937 remove unnecessary double allocation
BUG=

Review-Url: https://codereview.webrtc.org/2226933005
Cr-Commit-Position: refs/heads/master@{#13744}
2016-08-15 10:42:08 +00:00
0ccff57024 VoERTP_RTCP: Remove GetREDStatus and SetREDStatus
They always fail, because RED isn't supported.

BUG=webrtc:5922

Review-Url: https://codereview.webrtc.org/2055753002
Cr-Commit-Position: refs/heads/master@{#13743}
2016-08-15 10:34:52 +00:00
5bcc00e538 Changed folder structure in new mixer and fixed simple lint errors.
The folder structure is now as was agreed on in the 'Slim and Modular
WebRTC' effort.  Also added some dependencies that were previously in
another part of the tree.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2238803002
Cr-Commit-Position: refs/heads/master@{#13742}
2016-08-15 10:01:37 +00:00
714dd4e532 GN: Update tests to have the correct shard timeout value on Android.
TBR=mflodman@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2231413002
Cr-Commit-Position: refs/heads/master@{#13741}
2016-08-15 09:29:19 +00:00
5093b38497 Make variable for selecting if intervals without samples should be included in stats configurable (for rate counters).
BUG=

Review-Url: https://codereview.webrtc.org/2236923002
Cr-Commit-Position: refs/heads/master@{#13740}
2016-08-15 08:20:37 +00:00
c61ae74439 Roll chromium_revision 941118827f..7f405ec2b6 (411223:411933)
Change log: 941118827f..7f405ec2b6
Full diff: 941118827f..7f405ec2b6

Changed dependencies:
* src/buildtools: 33a32b8aa2..adb8bf4e8f
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/0d1b0961f9..96e1a25943
* src/third_party/usrsctp/usrsctplib: 9a3e5465e9..4ce5badc82
DEPS diff: 941118827f..7f405ec2b6/DEPS

No update to Clang.

TBR=
NOTRY=True

Review-Url: https://codereview.webrtc.org/2248563002
Cr-Commit-Position: refs/heads/master@{#13739}
2016-08-15 08:05:15 +00:00
414eb181d2 CQ: Temporarily disable iOS Simulator trybots
BUG=637666
TBR=ehmaldonado@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2244183002
Cr-Commit-Position: refs/heads/master@{#13738}
2016-08-15 07:44:09 +00:00
4cb5b64b16 Fix for data channels perpetually stuck in "closing" state.
If the data transport is destroyed while data is buffered (due to
the PC being closed, or a description set with data rejected), the
data channel was getting stuck in a "closing" state, waiting to
finish sending its buffered data. But since there's no more transport,
it will never get another chance to send buffered data.

It just needs to terminate non-gracefully and discard the buffered data
in this situation.

R=skvlad@webrtc.org, zhihuang@webrtc.org

Review URL: https://codereview.webrtc.org/2235843003 .

Cr-Commit-Position: refs/heads/master@{#13737}
2016-08-12 17:10:42 +00:00
64a7eab891 Update tests and DTX check for Opus 1.1.3.
DTX is now indicated by packets that may have a size of up to 2 bytes.
Ref: https://git.xiph.org/?p=opus.git;a=commit;h=1c311423c86b89eba27a494e17c79fefd7d75ab0

BUG=

Review-Url: https://codereview.webrtc.org/2158293003
Cr-Commit-Position: refs/heads/master@{#13736}
2016-08-12 11:36:14 +00:00
9591e3e82d Convert PeerConnectionTest to use the new capture APIs.
Review-Url: https://codereview.webrtc.org/2236323002
Cr-Commit-Position: refs/heads/master@{#13735}
2016-08-12 07:06:22 +00:00
62351c9923 Fixing problems with ICE candidate pair prioritization.
The main issue was that upon receiving a binding response with a srflx
mapped address attribute, the local candidate was not updated from local
to srflx. This means the two ICE agents view the same pair differently;
one sees it as "X<->srflx" while the other sees it as "local<->X". This
causes sub-optimal prioritization and could result in the wrong pair
being selected if using aggressive nomination.

The other issue was that TCP prflx candidates were not differentiated from
UDP prflx candidates. This lead to TCP prflx candidates prioritized above TCP
host candidates.

After fixing these issues, I was able to re-enable many disabled tests, as well
as restore the check for the candidate types of the controlled agent.

BUG=webrtc:1953,webrtc:2383
R=honghaiz@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2125823004 .

Cr-Commit-Position: refs/heads/master@{#13734}
2016-08-11 23:05:15 +00:00
6f82535f45 Enabling IPv6 socket recv timestamp test, and making less flaky.
The test worked by sleeping a certain time, then checking that the
difference between recv timestamps before and after the sleep was
within some margin of the requested sleep time.

However, this means that imprecision of SleepMs makes the test flaky.
This source of flakiness can be removed by comparing to the actual
time slept instead of the requested time.

Also making the margin larger, to further reduce the likelihood of
flakiness.

R=pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2111043004 .

Cr-Commit-Position: refs/heads/master@{#13733}
2016-08-11 22:38:40 +00:00
588783adcd Return nil from RTCPeerConnectionFactory when creation fails
RTCPeerConnectionFactory.createPeerConnection did not check the return
value of the native createPeerConnection() - so when the native PC
fails to be created, it could end up attempting to use a null pointer.

The change makes it return nil when the creation fails. The application
can then detect and respond to the failure.

Review-Url: https://codereview.webrtc.org/2240633004
Cr-Commit-Position: refs/heads/master@{#13732}
2016-08-11 21:29:32 +00:00
fe1ffb141b Remove unused SessionId from TransportChannel and PortAllocatorSession.
BUG=

Review-Url: https://codereview.webrtc.org/2237853002
Cr-Commit-Position: refs/heads/master@{#13731}
2016-08-11 19:37:51 +00:00
c8762a838f Remove StartSSLWithServer from SSLStreamAdapter.
It's not used by anything any more. We only use SSLStreamAdapter in
the mode where it verifies the peer's certificate using a signaled
digest.

R=pthatcher@webrtc.org, zhihuang@webrtc.org

Review URL: https://codereview.webrtc.org/2204883004 .

Cr-Commit-Position: refs/heads/master@{#13730}
2016-08-11 19:01:58 +00:00
f10976e2d0 Roll chromium_revision db8d32de07..941118827f (410624:411223)
Added third_party/ced to setup_links.py (needed for Android).

Change log: db8d32de07..941118827f
Full diff: db8d32de07..941118827f

Changed dependencies:
* src/third_party/ffmpeg: 4e878f7f64..75976ae026
* src/third_party/libFuzzer/src: 3ae6b1d110..764f3890a0
* src/third_party/libvpx/source/libvpx: 82070ae939..2d1e63d0c5
DEPS diff: db8d32de07..941118827f/DEPS

No update to Clang.

TBR=marpan@webrtc.org,ehmaldonado@webrtc.org,
BUG=
NOTRY=True

Review-Url: https://codereview.webrtc.org/2239673002
Cr-Commit-Position: refs/heads/master@{#13729}
2016-08-11 17:31:46 +00:00
3b74768152 Remove pbos@webrtc.org from WATCHLISTS.
BUG=
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/2238913003 .

Cr-Commit-Position: refs/heads/master@{#13728}
2016-08-11 16:52:21 +00:00
2e5cfcd6c2 Add periodic logging of video stats.
Add ToString method to: Call::Stats, VideoSendStream::Stats, VideoReceiveStream::Stats and log stats periodically (every 10 seconds).

BUG=

Review-Url: https://codereview.webrtc.org/2133073002
Cr-Commit-Position: refs/heads/master@{#13727}
2016-08-11 15:41:26 +00:00