generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
7,440 Commits 1 Branch 0 Tags 108 MiB
43c883954f
Commit Graph

7440 Commits

This Branch
This Branch
All Branches
Author SHA1 Message Date
sprang@webrtc.org
43c883954f Allow rtp packet history to dynamically expand in size. 
When using the paced sender, packets will be put into the rtp packet
history and then retreived from there again when it is time to send.

In some cases (low send bitrate and very large frames created) this
may overflow, causing packets to be overwritten in the packet history
before they have been sent.

Check this condition and expand history size if needed.

This is primarily triggered during screenshare, when
switching to a large picture with lots of high frequency
details in it.

BUG=4171
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34879004

Cr-Commit-Position: refs/heads/master@{#8195}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8195 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-29 09:09:41 +00:00
perkj@webrtc.org
827d7e806a Change AsyncInvoker to store its closure in a scoped_refptr instead of using a raw pointer. 
This is just a cosmetic change and does not solve a particular bug.

R=henrika@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38749004

Cr-Commit-Position: refs/heads/master@{#8194}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8194 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-29 08:54:17 +00:00
braveyao@webrtc.org
a742cb1f37 Enable DTLS for peerconnection example. If it's a loopback test, then we recreate another peerconnection with DTLS off. 
BUG=3872
TEST=Manual Test
R=jiayl@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36989004

Cr-Commit-Position: refs/heads/master@{#8193}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8193 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-29 04:23:39 +00:00
aluebs@webrtc.org
f17ee9c709 Add case to ApmTest.Process to test the extended filter mode 
R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40509004

Cr-Commit-Position: refs/heads/master@{#8192}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8192 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-29 00:04:18 +00:00
pkasting@chromium.org
e7a4a12f83 Add arraysize() macro from Chromium, and make use of it in a few places. 
This not only shortens some test code, it makes it more robust against changing
the lengths of the arrays later and forgetting to update the length constants
(which bit me).

BUG=none
TEST=none
R=hta@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34829004

Cr-Commit-Position: refs/heads/master@{#8191}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8191 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 21:37:13 +00:00
kjellander@webrtc.org
035e9123e9 Move channel_buffer.{h,cc} to common_audio. 
In https://code.google.com/p/webrtc/source/detail?r=8166
I added a check preventing GYP files from referencing
sources above their directory level.
This CL fixes the disallowed reference added in
https://code.google.com/p/webrtc/source/detail?r=8157
by moving channel_buffer.{h,cc} to common_audio for real.

BUG=4185
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35939004

Cr-Commit-Position: refs/heads/master@{#8190}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8190 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 19:57:44 +00:00
honghaiz@google.com
a67ca1a3bb Only report the first rtp packet because it indicates the media has started flowing. 
BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37829004

Cr-Commit-Position: refs/heads/master@{#8189}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8189 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 19:48:40 +00:00
guoweis@webrtc.org
a094cac11f Add stats for network merge. 
Currently, in ipc_network_manager.cc, the UMA WebRTC.PeerConnection.IPv4Interfaces and its IPv6
counter part counts the addresses, instead of the interfaces as when
chromium delivers available networks to WebRTC, each address is wrapped
inside an individual network object.

The plan is to replace the current MergeNetworkList with the new one once it's rolled into chromium.

BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36779004

Cr-Commit-Position: refs/heads/master@{#8188}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 19:34:17 +00:00
kjellander@webrtc.org
7d2b6a9346 Enable Clang warning implicit-fallthrough and annotate the code. 
BUG=4242
R=henrik.lundin@webrtc.org, stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34899004

Cr-Commit-Position: refs/heads/master@{#8187}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8187 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 18:38:13 +00:00
tommi@webrtc.org
a907e01c63 Adding constness. 
Make a few member variables in the Transport class officially const so that it's clear that locking isn't needed for access. There are getters for some of these (e.g. content_name()) that don't have locking or checking, so making the variables const is at least a way to guard against regressions. Also making the clock_ member in overuse_frame_detector.h const for clarity that it doesn't require a lock for access.

No code change.

Review URL: https://webrtc-codereview.appspot.com/35949004

Cr-Commit-Position: refs/heads/master@{#8186}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8186 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 17:33:45 +00:00
henrik.lundin@webrtc.org
664ccb7d8d Reland r8125: Modify some tests to never use DTX disable mode 
DTX disable mode will be removed as a part of the ACM redesign work.

This CL effectively reverts r8129, and relands r8125, but now using
assert instead of DCHECK.

COAUTHOR:kwiberg@webrtc.org
TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37839004

Cr-Commit-Position: refs/heads/master@{#8185}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8185 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 14:49:12 +00:00
asapersson@webrtc.org
37c0559c1e Notify jitter buffer about received FEC packets (to avoid sending NACK request for these packets). 
Don't copy codec specific header for empty packets in the jitter buffer.

BUG=3135
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37659004

Cr-Commit-Position: refs/heads/master@{#8184}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 13:58:40 +00:00
kjellander@webrtc.org
22c2f0572b Add "score" unit to SSIM perf score output. 
Currently, the SSIM values don't have a unit, which makes
them default to lower being better rather than the opposite
(which is the case for SSIM).

R=phoglund@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/41709004

Cr-Commit-Position: refs/heads/master@{#8183}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8183 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 13:52:37 +00:00
henrik.lundin@webrtc.org
4aecd008dd Add support for 40 and 60 ms frames to AudioEncoderIlbc 
BUG=3926
COAUTHOR:kwiberg@webrtc.org

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37789004

Cr-Commit-Position: refs/heads/master@{#8182}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8182 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 13:16:44 +00:00
sprang@webrtc.org
2a6558c2a5 Make sure ByteReader<T>::Read* is properly constified. 
Also, start using it in real code...

BUG=
R=holmer@google.com, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37809004

Cr-Commit-Position: refs/heads/master@{#8181}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8181 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 12:38:16 +00:00
kjellander@webrtc.org
7aef80c6d1 GN: Remove webrtc_base target in favor for rtc_base. 
The last reference to the old target name was
removed in https://crrev.com/7c9149860a8a0ca24350d2e80dbc280990a0cbb7

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33079004

Cr-Commit-Position: refs/heads/master@{#8179}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8179 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 07:55:45 +00:00
marpan@webrtc.org
9b64a6edd7 Adjust parameter in videoprocessor_integrationtest for VP9. 
TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/35919004

Cr-Commit-Position: refs/heads/master@{#8178}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8178 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 23:59:16 +00:00
marpan@webrtc.org
dc8a9da386 Adjust qp-max settinhg in VP9 wrapper. 
More closely matches the qp-max setting used in VP8.

TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/39709004

Cr-Commit-Position: refs/heads/master@{#8177}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8177 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 23:08:39 +00:00
andrew@webrtc.org
922cfcd150 Use non-zero data in AudioRingBufferTest. 
TBR=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35909004

Cr-Commit-Position: refs/heads/master@{#8176}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8176 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 21:59:44 +00:00
tkchin@webrtc.org
36401aba62 Update GAE API paths for join/leave. 
BUG=4221
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33069004

Cr-Commit-Position: refs/heads/master@{#8174}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8174 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 21:35:16 +00:00
henrik.lundin@webrtc.org
8bb32d600b Minor updates to AudioEncoderCng 
Removing sample_rate_hz_ from AudioEncoderCng and from the config
struct. The sample rate will now be read from the underlying speech
codec.

BUG=3926
COAUTHOR:kwiberg@webrtc.org

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40559004

Cr-Commit-Position: refs/heads/master@{#8173}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8173 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 20:54:22 +00:00
tnakamura@webrtc.org
db1ebf6c0c Add jakehilton@gmail.com to AUTHORS 
BUG=3918
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34889004

Cr-Commit-Position: refs/heads/master@{#8172}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8172 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 19:15:48 +00:00
henrik.lundin@webrtc.org
478cedc055 Add new methods to AudioEncoder interface 
The following three methods are added:
rtp_timestamp_rate_hz()
SetTargetBitrate()
SetProjectedPacketLossRate()

Default implementations are provided, and a few overrides are
implemented. AudioEncoderCopyRed and AudioEncoderCng propagate the new
methods to the underlying speech codec.

BUG=3926
COAUTHOR:kwiberg@webrtc.org

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34049004

Cr-Commit-Position: refs/heads/master@{#8171}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8171 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 18:25:40 +00:00
bjornv@webrtc.org
5614cf16e7 audio_processing: Use fixed aggregation window in delay metrics 
Previously, the delay estimate history was reset every time the metrics were pulled. This required all clients to be on the same thread and make use of one call.

Now we use a fixed aggregation window of one second and when a client pulls the metrics you get the latest value.
Under certain circumstances like tests you would like to have the aggregation window set to the recording length. We therefore turn on the fixed aggregation window after the first call.

BUG=2994
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38759004

Cr-Commit-Position: refs/heads/master@{#8170}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8170 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 18:10:27 +00:00
kjellander@webrtc.org
6e251822cd Whitespace change after enabling gnumbd 
TBR=machenbach@chromium.org

Review URL: https://webrtc-codereview.appspot.com/37019004

Cr-Commit-Position: refs/heads/master@{#8169}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8169 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 16:46:13 +00:00
kjellander@webrtc.org
ccd608eeab Whitespace change for git updater 
TBR=machenbach@chromium.org

Review URL: https://webrtc-codereview.appspot.com/41669004

Cr-Commit-Position: refs/heads/master@{#8168}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8168 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 16:37:48 +00:00
kjellander@webrtc.org
0bc73a1b72 Whitespace change to trigger git updater 
TBR=machenbach@chromium.org

Review URL: https://webrtc-codereview.appspot.com/34869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8167 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 14:13:13 +00:00
kjellander@webrtc.org
f68ffca050 Add PRESUBMIT check for GYP files including source files above itself. 
This is needed because some tools does not support files
located above the project generated.

BUG=4185
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8166 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 13:13:24 +00:00
kjellander@webrtc.org
76e5e207ad Roll chromium_revision 4664fe0..9070a80 (312733:313233) 
Relevant changes:
* src/third_party/boringssl/src: 5fa3eba..347f025
* src/third_party/libvpx: 8dc6ea9..5da40ca
* src/tools/gyp: adb7d24..b28bd7d
* src/tools/swarming_client: e98dde9..d863df3
Details: 4664fe0..9070a80/DEPS

Clang version was not updated in this roll.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8165 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 13:11:10 +00:00
asapersson@webrtc.org
273fbbb921 Update StreamDataCounter with FEC bytes. 
Add histograms stats for send/receive FEC bitrate:
- "WebRTC.Video.FecBitrateReceivedInKbps"
- "WebRTC.Video.FecBitrateSentInKbps"

Correct media payload bytes in StreamDataCounter to not include FEC bytes.

Fix stats for rtcp packets sent/received per minute (regression from r7910).

BUG=crbug/419657
R=holmer@google.com, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 12:17:29 +00:00
bjornv@webrtc.org
70117a83d4 AEC: Implements a new function for calculating delay metrics 
Two new member variables have been added and the code for calculating the delay metrics have been moved to a function.

BUG=2994
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8163 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 11:30:54 +00:00
magjed@webrtc.org
fc5ad95fec Reland of: "Implement elapsed time and capture start NTP time estimation." revision @8139 
Link to original CL: https://review.webrtc.org/36909004/

R=pbos@webrtc.org
TBR=pthatcher@webrtc.org
BUG=4227

Review URL: https://webrtc-codereview.appspot.com/39669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8162 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 09:57:01 +00:00
glaznev@webrtc.org
8501ee632b Support VP8 HW decoding on devices with Exynos codec. 
R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8160 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 23:07:19 +00:00
pkasting@chromium.org
df9a41d270 Fix bug in GetREDStatus(): it doesn't actually return the current status. 
BUG=none
TEST=none
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8159 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 22:35:29 +00:00
glaznev@webrtc.org
82415e395f Update AppRTCDemo to use renamed GAE messages. 
BUG=4221
R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8158 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 22:22:50 +00:00
andrew@webrtc.org
041035b390 Add an AudioRingBuffer class wrapper for the ring_buffer.h C interface. 
Integrate it in Blocker to demonstrate use.

TEST=beamforming sounds good.
R=aluebs@webrtc.org, mgraczyk@chromium.org, sahark@google.com

Review URL: https://webrtc-codereview.appspot.com/36799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8157 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 21:23:53 +00:00
pkasting@chromium.org
4dba2e98a2 Consolidate anonymous namespace content and file-static methods to all be in the 
anonymous namespace, in preparation for refactoring a few of the functions a
little.

No code change.

BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8155 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 19:59:32 +00:00
kjellander@webrtc.org
d7e34e1086 Make it easier to use external libyuv + cleanup GYP files. 
It is now easier to use an external libyuv library.
Fix some GYP errors.
Remove the temporary webrtc_base target (depends on
https://codereview.chromium.org/865603002/ being landed
first).

BUG=4185
R=andresp@webrtc.org, andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8154 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 19:17:26 +00:00
bjornv@webrtc.org
d25c034051 Refactor common_audio/vad: Removed usage of macro WEBRTC_SPL_MUL_16_16() 
BUG=3348,3353
TESTED=Locally on Mac and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8152 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 15:32:47 +00:00
tommi@webrtc.org
04cd466bd5 Move ThreadChecker into rtc_base_approved. 
To do this, I'm removing ThreadChecker's dependency on the 'Thread' class, so that the checker works with any thread and doesn't rely on TLS.
Also simplifying CriticalSection's implementation on Windows since a critical section on Windows already knows what thread currently owns the lock.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8151 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 15:27:29 +00:00
marpan@webrtc.org
38d11b8529 Enable encoder multi-threading for VP9. 
R=stefan@webrtc.org
TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/41489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8150 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 15:21:36 +00:00
kwiberg@webrtc.org
6f200b5b87 Temporarily revert r8147 ("Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h") 
Some out-of-tree code that uses base/scoped_ptr.h is defining nullptr
to 0, which causes an obvious compilation error and perhaps other
subtle problems. I'm hoping to get that sorted out and re-land this CL
soon.

TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8149 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 13:03:32 +00:00
henrik.lundin@webrtc.org
b6fab2b1cd Introduce rtc::CheckedDivExact 
Use the new method to replace local ones in AudioEncoder{Opus,Isac}.

COAUTHOR:kwiberg@webrtc.org

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8148 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 11:08:53 +00:00
kwiberg@webrtc.org
19eb4e4b86 Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h 
The latter file was more up-to-date. The files are now identical
with the following exceptions:

  * The namespace used (rtc vs. webrtc).

  * The name of the include guard.

  * base/scoped_ptr.h still has two extra methods, accept() and use().

  * base/scoped_ptr.h still includes webrtc/base/common.h even though
    it doesn't need it itself, since several .cc files expect to get
    it for free by incuding base/scoped_ptr.h. This is of course bad
    manners, and the "unused" include will be removed in a future CL.

A later CL will remove system_wrappers/interface/scoped_ptr.h.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8147 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 08:57:57 +00:00
kjellander@webrtc.org
995b4c9e8a Remove win_asan trybot from PRESUBMIT.py 
Removing it since it no longer exists.
See https://codereview.chromium.org/872263002/

TBR=phoglund@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/36979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8146 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-25 19:27:03 +00:00
kjellander@webrtc.org
acb8085678 Roll chromium_revision c086b4e..4664fe0 (312108:312733) 
Mainly to pick up the MIPS changes in
https://codereview.chromium.org/843563002/
for which the changes in
https://webrtc-codereview.appspot.com/41399004/
are included in this CL.

Relevant changes:
* src/third_party/android_tools: 56b3d3e..aaeda3d
* src/third_party/boringssl/src: ca9a538..5fa3eba
* src/third_party/libvpx: 4f9bd1b..8dc6ea9
* src/third_party/openmax_dl: 1a4171c..8f7bf0b
* src/tools/gyp: 194ec65..adb7d24
* src/tools/swarming_client: 0a795bd..e98dde9
Details: c086b4e..4664fe0/DEPS

Clang version was not updated in this roll.

BUG=4214, 4222
TBR=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8145 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-25 19:17:56 +00:00
tkchin@webrtc.org
7519de519e Revert 8136 "Remove frame copy in ViEExternalRendererImpl::Rende..." 
> Remove frame copy in ViEExternalRendererImpl::RenderFrame
> 
> Add new interface for delivering frames to ExternalRenderer. The purpose is to avoid having to extract a packed buffer from I420VideoFrame, which will cause a deep frame copy.
> 
> BUG=1128
> R=mflodman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/36489004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8144 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-23 21:20:41 +00:00
tkchin@webrtc.org
0f98844749 Revert 8139 "Implement elapsed time and capture start NTP time e..." 
> Implement elapsed time and capture start NTP time estimation.
> 
> These two elements are required for end-to-end delay estimation.
> 
> BUG=1788
> R=stefan@webrtc.org
> TBR=pthatcher@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/36909004

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8143 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-23 21:17:38 +00:00
jiayl@webrtc.org
dacdd9403d Reland r7980: 
Accept incoming pings before remote answer is set, to reduce connection latency.
Set ICE connection state to 'checking' after setting the remote answer, so that it can transition into 'connected' if the peer reflexive connection is up before any remote candidate is set. See more details in crbug/446908

BUG=4068, crbug/446908
R=juberti@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8141 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-23 17:33:34 +00:00
fdegans@chromium.org
8919cfe9ce Change a GYP reference to cpufeatures.gypi 
This will allow us to move the remaining GYP file in android_tools
to the chromium repository by removing the direct reference to it.

BUG=webrtc:4115
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8140 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-23 16:35:17 +00:00