Commit Graph

2162 Commits

Author SHA1 Message Date
5082c83320 Make type and constructors in EglBase14 public.
This type is used for GL contexts in android hardware encode/decode, so
it seems correct/useful to make it publicly constructable.

Review URL: https://codereview.webrtc.org/1604403002

Cr-Commit-Position: refs/heads/master@{#11331}
2016-01-21 10:13:00 +00:00
d26fadb454 Delete GetRenderer method, used only by the tests.
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1602283003

Cr-Commit-Position: refs/heads/master@{#11328}
2016-01-21 07:26:44 +00:00
057ecf01e4 Making WebRtcSession fire a destroyed signal.
This ensures the DtmfSender won't try to access it after it's
destroyed.

BUG=webrtc:5403

Review URL: https://codereview.webrtc.org/1590333004

Cr-Commit-Position: refs/heads/master@{#11327}
2016-01-20 22:30:50 +00:00
1d61a518ac Send key frame if time difference between incoming frames exceeds a certain limit.
BUG=

Review URL: https://codereview.webrtc.org/1510913007

Cr-Commit-Position: refs/heads/master@{#11315}
2016-01-20 09:13:51 +00:00
8a2c31d208 Make it possible to run peerconnection_unittests on Android.
- renamed libjingle_peerconnection_unittest to peerconnection_unittest to circumvent cr issue http://crbug.com/543820

TEST=On an android build webrtc/build/android/test_runner.py gtest -s peerconnection_unittests --verbose -t 900
BUG=webrtc:2365,543820
NOTRY=True

Review URL: https://codereview.webrtc.org/1602443004

Cr-Commit-Position: refs/heads/master@{#11304}
2016-01-19 14:20:07 +00:00
c4c8485662 Deleted renderer-related SetSize methods, and all uses.
webrtc::VideoRendererInterface::SetSize was completely unused.

cricket::VideoRenderer::SetSize only had dummy implementations
returning true and doing nothing.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1574963002

Cr-Commit-Position: refs/heads/master@{#11298}
2016-01-19 08:52:55 +00:00
81354f5ef6 Added mute logic to VideoTrackRenderers.
If the track is disabled, replace incoming frames by black frames. Affects local rendering of disabled tracks.
Also intended to replace the similar logic in
WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame, once
VideoRtpSender is hooked up as a renderer for the track.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1575223003

Cr-Commit-Position: refs/heads/master@{#11297}
2016-01-19 08:23:31 +00:00
f5a3a93d26 Add 5-argument wrapper WebRtcVideoFrame::InitToBlack
For some reason, inheriting it doesn't work.

BUG=webrtc:5426
R=perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1602543003 .

Cr-Commit-Position: refs/heads/master@{#11290}
2016-01-18 14:24:54 +00:00
8b1e431231 Delete remnants of non-square pixel support from cricket::VideoFrame.
If ever needed, add some aspect ratio parameter, without pixel_width
and pixel_height arguments cluttering commonly used functions.

BUG=webrtc:5426

Committed: https://crrev.com/709513d4133107d5c02aed34a5ee99444c4d4e25
Cr-Commit-Position: refs/heads/master@{#11243}

Review URL: https://codereview.webrtc.org/1586613002

Cr-Commit-Position: refs/heads/master@{#11287}
2016-01-18 09:46:33 +00:00
cec0a08275 Add a new interface for creating a udp socket in which it binds the socket to a network if the network handle is set.
Plus, in stunport, turnport and allocation sequence, create a socket using the new interface.

BUG=

Review URL: https://codereview.webrtc.org/1556743002

Cr-Commit-Position: refs/heads/master@{#11279}
2016-01-15 22:49:15 +00:00
f4decb5469 Add QP statistics logging to Android HW encoder.
- Similar to HW decoder log timing for first 5 frames
of HW encoder input and output to track correct encoder
start.
- Increment input timestamp when frame is dropped.
- Better calculate average encoding time.
- Use different tags for encoder and decoder modules.

BUG=b/26504665

Review URL: https://codereview.webrtc.org/1587943004

Cr-Commit-Position: refs/heads/master@{#11277}
2016-01-15 21:49:26 +00:00
884f58523a Storing raw audio sink for default audio track.
BUG=webrtc:5250

Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99
Cr-Commit-Position: refs/heads/master@{#11230}

Review URL: https://codereview.webrtc.org/1551813002

Cr-Commit-Position: refs/heads/master@{#11275}
2016-01-15 17:20:08 +00:00
79a5a83e10 Adapt to boringssl's new defaults.
This is now a merge with patchset #2 of https://codereview.webrtc.org/1550773002 after that CL was reverted.

BUG=webrtc:5381

Review URL: https://codereview.webrtc.org/1589493004

Cr-Commit-Position: refs/heads/master@{#11273}
2016-01-15 15:16:54 +00:00
d66b44d565 Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
Original review: https://codereview.webrtc.org/1413483003/

The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.

TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
BUG=webrtc:4741
Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a
Cr-Commit-Position: refs/heads/master@{#11093}

Review URL: https://codereview.webrtc.org/1540103002

Cr-Commit-Position: refs/heads/master@{#11267}
2016-01-15 11:06:41 +00:00
0f7d2939e0 Revert changes to default option setting in https://codereview.webrtc.org/1500633002/
As found by aluebs@, the changes breaks ability to create AecDumps: https://codereview.webrtc.org/1530333007/

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1568853002

Cr-Commit-Position: refs/heads/master@{#11265}
2016-01-15 09:40:45 +00:00
dc305db059 Add ApplyPacketOptions()
When libjingle is compied with ENABLE_EXTERNAL_AUTH the sending socket
needs to update RTP header in order for the outgoing packet to be
valid. The corresponding code was in chromium in
content/browser/renderer_host/p2p/socket_host.cc and it was impossible
to reuse it anywhere else. This CL moves this code to
talk/media/base/rtputils.h/cc, so it can be used outside of chrome.

BUG=crbug.com/547158
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1578323002 .

Cr-Commit-Position: refs/heads/master@{#11261}
2016-01-15 01:15:05 +00:00
20ac434010 Fix a test bot failure.
That was caused by https://codereview.webrtc.org/1581903002/

BUG=
R=pthatcher@google.com
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1587373002 .

Cr-Commit-Position: refs/heads/master@{#11260}
2016-01-14 23:56:35 +00:00
e1f9d837ae Adding AddTrack/RemoveTrack to native PeerConnection API.
Also, now creating the RtpSender/RtpReceiver proxy objects immediately,
rather than waiting until when GetSenders/GetReceivers is called.

Review URL: https://codereview.webrtc.org/1563403002

Cr-Commit-Position: refs/heads/master@{#11259}
2016-01-14 23:35:46 +00:00
67b1e1ab0b Put options as the argument of the java PeerConnectionFactory constructor.
BUG=

Review URL: https://codereview.webrtc.org/1581903002

Cr-Commit-Position: refs/heads/master@{#11257}
2016-01-14 22:45:44 +00:00
fcfc804e43 Eliminate defines in talk/
Replace LINUX, OSX and IOS defines with WEBRTC_ prefixed versions.
Remove no longer used defines from talk/build/common.gypi due to
previously migrated sources (into webrtc/p2p and webrtc/libjingle).

When this is rolled into Chromium, we can also clean up the platform
defines in
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle.gyp

NOTRY=True
BUG=webrtc:5420
TESTED=Ran all compile trybots with --clobber flag.
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1588453005

Cr-Commit-Position: refs/heads/master@{#11254}
2016-01-14 19:01:25 +00:00
3542013f58 Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
Reason for revert:
We're getting boringssl version conflicts. Reverting for now.

Original issue's description:
> Update with new default boringssl no-aes cipher suites. Re-enable tests.
>
> This undoes https://codereview.webrtc.org/1533253002 (except the DEPS part).
>
> BUG=webrtc:5381
> R=davidben@webrtc.org, henrika@webrtc.org
>
> Committed: https://crrev.com/31c8d2eac5aec977f584ab0ae5a1d457d674f101
> Cr-Commit-Position: refs/heads/master@{#11250}

TBR=davidben@webrtc.org,henrika@webrtc.org,torbjorng@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5381

Review URL: https://codereview.webrtc.org/1586183002

Cr-Commit-Position: refs/heads/master@{#11253}
2016-01-14 17:14:06 +00:00
31c8d2eac5 Update with new default boringssl no-aes cipher suites. Re-enable tests.
This undoes https://codereview.webrtc.org/1533253002 (except the DEPS part).

BUG=webrtc:5381
R=davidben@webrtc.org, henrika@webrtc.org

Review URL: https://codereview.webrtc.org/1550773002 .

Cr-Commit-Position: refs/heads/master@{#11250}
2016-01-14 14:18:02 +00:00
688e308a35 Re-land: "Use an explicit identifier in Config"
This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.

Original CL: https://codereview.webrtc.org/1538643004/

TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1589573004

Cr-Commit-Position: refs/heads/master@{#11248}
2016-01-14 12:32:51 +00:00
268493a96b Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
Reason for revert:
These changes broke chrome.

Need to temporarily keep methods InitToEmptyBuffer, InitToBlack, CreateEmptyFrame with old but ignored arguments for pixel_width and pixel_height. Then update chrome, and delete the old methods in a separate cl.

Original issue's description:
> Delete remnants of non-square pixel support from cricket::VideoFrame.
>
> If ever needed, add some aspect ratio parameter, without pixel_width
> and pixel_height arguments cluttering commonly used functions.
>
> BUG=webrtc:5426
>
> Committed: https://crrev.com/709513d4133107d5c02aed34a5ee99444c4d4e25
> Cr-Commit-Position: refs/heads/master@{#11243}

TBR=pthatcher@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1583223002

Cr-Commit-Position: refs/heads/master@{#11246}
2016-01-14 10:35:30 +00:00
709513d413 Delete remnants of non-square pixel support from cricket::VideoFrame.
If ever needed, add some aspect ratio parameter, without pixel_width
and pixel_height arguments cluttering commonly used functions.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1586613002

Cr-Commit-Position: refs/heads/master@{#11243}
2016-01-14 07:43:56 +00:00
2d110be77f Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
Reason for revert:
tommi pointed out that using a refptr for the sink may cause issues. Will reland with a slightly different approach.

Original issue's description:
> Storing raw audio sink for default audio track.
>
> BUG=webrtc:5250
>
> Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99
> Cr-Commit-Position: refs/heads/master@{#11230}

TBR=pthatcher@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5250

Review URL: https://codereview.webrtc.org/1588693002

Cr-Commit-Position: refs/heads/master@{#11241}
2016-01-13 20:00:29 +00:00
fca54f41ad Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
Reason for revert:
Reverting due to problem with roll:

/b/build/slave/linux/build/src/buildtools/linux64/gn gen //out/Release '--args=ffmpeg_branding="Chrome" proprietary_codecs=true is_debug=false is_component_build=false use_goma=true goma_dir="/b/build/goma" symbol_level=1 dcheck_always_on=true' --check --runtime-deps-list-file=/b/build/slave/linux/build/src/out/Release/runtime_deps
  -> returned 1
ERROR at //third_party/webrtc/BUILD.gn:245:18: Item not found
    configs -= [ "//build/config/clang:find_bad_constructs" ]
                 ^-----------------------------------------
You were trying to remove "//build/config/clang:find_bad_constructs"
from the list but it wasn't there.
GN gen failed: 1
step returned non-zero exit code: 1
@@@STEP_FAILURE@@@

Original issue's description:
> Use an explicit identifier in Config
>
> This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.
>
> Committed: https://crrev.com/25249d92d3cf105bcc7b684c8924ccdbc9afcb93
> Cr-Commit-Position: refs/heads/master@{#11231}

TBR=henrik.lundin@webrtc.org,stefan@webrtc.org,tommi@chromium.org,aluebs@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/1586563003

Cr-Commit-Position: refs/heads/master@{#11239}
2016-01-13 16:12:07 +00:00
306efadffa Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
BUG=webrtc:4963
TBR=pbos@webrtc.org
NOTRY=True

Review URL: https://codereview.webrtc.org/1577233005

Cr-Commit-Position: refs/heads/master@{#11237}
2016-01-13 15:51:32 +00:00
25249d92d3 Use an explicit identifier in Config
This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.

Review URL: https://codereview.webrtc.org/1538643004

Cr-Commit-Position: refs/heads/master@{#11231}
2016-01-13 02:50:31 +00:00
e591f9377f Storing raw audio sink for default audio track.
BUG=webrtc:5250

Review URL: https://codereview.webrtc.org/1551813002

Cr-Commit-Position: refs/heads/master@{#11230}
2016-01-13 00:45:33 +00:00
6955870806 Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1316523002 .

Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-13 00:26:55 +00:00
3e1cfa7edb Delete unused method webrtc::VideoRendererInterface::SetSize.
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1582493002

Cr-Commit-Position: refs/heads/master@{#11223}
2016-01-12 14:39:25 +00:00
127782bbb1 Add default dummy implementation of cricket::VideoRenderer::SetSize, to easy later removal.
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1581583002

Cr-Commit-Position: refs/heads/master@{#11218}
2016-01-12 11:39:20 +00:00
b2328d11dc Remove additional channel constraints when Beamforming is enabled in AudioProcessing
The general constraints on number of channels for AudioProcessing is:
num_in_channels == num_out_channels || num_out_channels == 1

When Beamforming is enabled and additional constraint was added forcing:
num_out_channels == 1

This artificial constraint was removed by adding upmixing support in CopyTo, since it was already supported for the AudioFrame interface using InterleaveTo.

Review URL: https://codereview.webrtc.org/1571013002

Cr-Commit-Position: refs/heads/master@{#11215}
2016-01-12 04:32:32 +00:00
a7446d2a50 Change DTLS default from 1.0 to 1.2 for webrtc.
This changes for standalone webrtc applications.

BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1548733002 .

Cr-Commit-Position: refs/heads/master@{#11211}
2016-01-11 23:27:12 +00:00
27ed3cc28c SCTP: Stopped accepting SSRCs higher than max.
Seems to fix asan-related crash.

BUG=https://code.google.com/p/chromium/issues/detail?id=570261

Review URL: https://codereview.webrtc.org/1571853002

Cr-Commit-Position: refs/heads/master@{#11205}
2016-01-11 18:24:35 +00:00
f475d365a2 Properly handle different transports having different SSL roles.
This meant splitting "transport_options" into audio/video/data options,
for when creating the answer, and giving "GetSslRole" a "transport_name"
parameter so we can retrieve the current role on a per-transport basis.

BUG=webrtc:4525
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1516993002 .

Cr-Commit-Position: refs/heads/master@{#11192}
2016-01-08 23:36:06 +00:00
25702cb162 Misc. small cleanups.
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests).  Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks

All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.

BUG=none
TEST=none

Review URL: https://codereview.webrtc.org/1534193008

Cr-Commit-Position: refs/heads/master@{#11191}
2016-01-08 21:50:32 +00:00
37ebcf0ce5 Reland "Add APK targets to build libjingle tests for Android."
patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/

This reverts commit bc14164aad254e72ce4d1e381b912b7d3acf5391.

We have made more preparations downstream, so this should work now. Original CL by perkj@.

BUG=webrtc:2365
The work started from the work by kjellander@ in https://codereview.webrtc.org/1413663003/

Review URL: https://codereview.webrtc.org/1570513004

Cr-Commit-Position: refs/heads/master@{#11186}
2016-01-08 13:05:01 +00:00
fbeb97e01f Fix clang warning in peerconnection_jni.cc
TEST= export GYP_DEFINES="OS=android clang=1" ...
      ninja -C out/Debug AppRTCDemo
BUG=webrtc:5399

Review URL: https://codereview.webrtc.org/1561073005

Cr-Commit-Position: refs/heads/master@{#11181}
2016-01-08 08:43:15 +00:00
893505d0fb Adding unit test to ensure TURN server priorities are unique.
BUG=webrtc:5209
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1570563002 .

Cr-Commit-Position: refs/heads/master@{#11177}
2016-01-07 23:12:53 +00:00
e5ba13bc09 Adding a way for a Java RtpSender to set a track without taking ownership.
This means that the track will still have a reference count after the
PeerConnection and RtpSender have been destroyed.

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

Review URL: https://codereview.webrtc.org/1566103003 .

Cr-Commit-Position: refs/heads/master@{#11176}
2016-01-07 23:11:33 +00:00
13f61dfea5 Move fake-handle frame creation into test target.
Renames CreateFakeNativeHandleFrame to FakeNativeHandle::CreateFrame and
moves into test.gyp target 'fake_video_frames' which contains previous
frame_generator target.

Removes unused warnings from includers of
webrtc/test/fake_texture_frame.h which did not use the function above.

BUG=webrtc:5398
R=kjellander@webrtc.org
TBR=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1554223002 .

Cr-Commit-Position: refs/heads/master@{#11149}
2016-01-04 21:36:49 +00:00
60ca31bf5d Roll chromium_revision d66326c..4df108a (367167:367307)
The changes in d66326c..4df108a/build/common.gypi
enables a lot more warnings, which have been disabled/fixed in this CL.
See tracking bugs for remaining work.

Change log: d66326c..4df108a
Full diff: d66326c..4df108a

Changed dependencies:
* src/buildtools: fee7f1e..6d0c448
* src/third_party/libsrtp: b8dd754..8a7662a
DEPS diff: d66326c..4df108a/DEPS

No update to Clang.

BUG=webrtc:5397, webrtc:5398, webrtc:5399
TBR=hta@webrtc.org, perkj@webrtc.org
NOTRY=True

Review URL: https://codereview.webrtc.org/1553033002

Cr-Commit-Position: refs/heads/master@{#11147}
2016-01-04 18:16:01 +00:00
0c7e9f540b Removing webrtc::PortAllocatorFactoryInterface.
ICE servers are now passed directly into PortAllocator,
making PortAllocatorFactoryInterface redundant. This CL also
moves SetNetworkIgnoreMask to PortAllocator.

R=phoglund@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1520963002 .

Cr-Commit-Position: refs/heads/master@{#11139}
2015-12-29 22:15:02 +00:00
3f7219be70 Fixing issue where description contains empty ICE ufrag/pwd.
The issue occurred when deserializing and then serializing a rejected
content description, which doesn't have the ICE ufrag/pwd in the first
place.

BUG=webrtc:5105

Review URL: https://codereview.webrtc.org/1534363002

Cr-Commit-Position: refs/heads/master@{#11134}
2015-12-28 23:17:22 +00:00
e6bf587259 Deleted VideoCapturer::screencast_max_pixels, together with
VideoChannel::GetScreencastMaxPixels and VideoChannel::GetScreencastFps.

Unused in webrtc, also unused in everything indexed by google and chromium code search. With the exception of the magicflute plugin, which I'm told doesn't matter.

Review URL: https://codereview.webrtc.org/1532133002

Cr-Commit-Position: refs/heads/master@{#11108}
2015-12-21 21:18:18 +00:00
2f042f26a3 Roll chromium_revision 1b6c421..db567a8 (365999:366304)
I had to disable some Dtls12Both tests failing under MSan (see bug).
Notice those errors started happening in the range of
https://boringssl.googlesource.com/boringssl.git/+log/afd565f..9f897b2
while this CL brings in an even newer BoringSSL (that still has the same problem).

Change log: 1b6c421..db567a8
Full diff: 1b6c421..db567a8

Changed dependencies:
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/afd565f..afe57cb
* src/third_party/libyuv: 1019e45..1ccbf8f
* src/third_party/nss: a676aa0..aee1b12
DEPS diff: 1b6c421..db567a8/DEPS

No update to Clang.

NOTRY=True
BUG=webrtc:5381
TBR=torbjorng@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1533253002

Cr-Commit-Position: refs/heads/master@{#11095}
2015-12-20 20:25:17 +00:00
a4df27b671 Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )
Reason for revert:
Compile error on Android needs to be fixed before relanding.

Original issue's description:
> Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
>
> The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
> Original review: https://codereview.webrtc.org/1413483003/
>
> The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.
>
> NOTRY=true
> TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
> BUG=webrtc:4741
>
> Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a
> Cr-Commit-Position: refs/heads/master@{#11093}

TBR=glaznev@webrtc.org,henrik.lundin@webrtc.org,solenberg@google.com,henrikg@webrtc.org,perkj@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1537213002

Cr-Commit-Position: refs/heads/master@{#11094}
2015-12-19 18:14:18 +00:00
f4f5cb0927 Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
Original review: https://codereview.webrtc.org/1413483003/

The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.

NOTRY=true
TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1541633002

Cr-Commit-Position: refs/heads/master@{#11093}
2015-12-19 18:02:39 +00:00