| Age | Commit message (Collapse) | Author |
|---|
|
am: 3d524127ba
Change-Id: Ief1ef735ecb67f9b80219eb65432851b4e85dc64
|
|
Native tests are also moved.
Test: mma -j BOARD_VNDK_VERSION=current
Test: m -j -k BOARD_VNDK_VERSION=current has no errors for webrtc
Bug: 62489821
Change-Id: I7f182403c2dabafc20b53f1046f68c1af81cede2
|
|
am: f08d623d4e
Change-Id: I1e376149ca4c357b34538e21b4673ca4d788d876
|
|
am: 83cb4ee1d7
Change-Id: Ia8eae124a9a4a6d13740f8ceb4d32198dbfbb296
|
|
Fix a missing dependency.
Test: mma -j
Test: m -j -k BOARD_VNDK_VERSION=current has no error for webrtc
Bug: 62489821
Merged-In: I479a4addb59b42dd86cc5dfa02cd4ff6d1b6d626
Change-Id: I479a4addb59b42dd86cc5dfa02cd4ff6d1b6d626
(cherry picked from commit c6f833ab1ab6ccb6e0401fa61a182c516b849d87)
|
|
Native tests are not moved. Tests include:
webrtc_apm_process_test
webrtc_isac_test
webrtc_apm_unit_test
Test: mma -j
Test: m -j -k BOARD_VNDK_VERSION=current has no errors for webrtc
Bug: 62489821
Merged-In: I013303de263866cbf368f3f89327c5357f9cecdb
Change-Id: I013303de263866cbf368f3f89327c5357f9cecdb
(cherry picked from commit 8df7e85368569b7cd0afc1ce231b8b3a0ab49333)
|
|
SSL_CTX_set_max_version." am: 0d85cc3c3c am: 1335ab5cae
am: c0c6db4af3
Change-Id: Ie40178c8e1e8c2c9e13401142a202d9c61e8cbc1
|
|
|
|
These functions are identical. BoringSSL added these APIs, then OpenSSL
1.1.0 added similar ones but with slightly longer names. We're
standardizing on the OpenSSL names to avoid API skew.
This is a cherry-pick of https://codereview.webrtc.org/2550423004.
Test: mma
Change-Id: Iffe4416bd300d71bbaa94c7c127ae89d6286cd7b
|
|
am: 564d8d32cb
Change-Id: Ic722d6421a581dc727d3f376a10a282091e3a197
|
|
* Modeled after Elliott Hughes' commit here: I82fb692c758d4855efa12ca7b743ce2893776a77
* Allows for compatibility with ARMv8
Test: mm
Change-Id: Ic80b0ff9416c781efed1b9686b6dff077c75e673
|
|
am: d1d58ca589
Change-Id: Ib8e42e572f0c64557b4412b18584260be647f55f
|
|
We should not copy OWNERS files from upstream,
or the owners should be registered in Gerrit Code Review.
Bug: 33166666
Test: default build targets
Change-Id: Ibfd47e643f03678bb65880653383adb84809169d
|
|
Test: builds without warnings
Change-Id: Ic88c90167584c97689ac32e0c03d692560e14570
|
|
am: c55415d27b
Change-Id: Ia1b9be2f0ccc56ead8f3efb35f11a667abc80d2f
|
|
This fixes x86+arm multilib build.
Bug: 27526885
Test: Build system refactoring CL. Existing unit tests still pass.
Change-Id: I7290198a3671af3688cf125ffb1e6979344c0a8d
|
|
Test: builds frameworks/av with no warnings
Change-Id: I9efb1c6d532a8be5742e5b16a4787a7e536f00cf
|
|
Suppress warnings until upstream can fix them.
BUG: 27074506
Change-Id: If7e6f190100fba025d25d2634d1c9a657cc24854
|
|
* git merge 04cb763
* See all upstream changes since the previous merge in branch aosp/upstream-master: git diff cb3f9bd..04cb763
* Modify webrtc/.gitignore to keep *.mk files.
* Removed old files from *.mk files:
- thread.cc
- thread_posix.cc
* Add new files to *.mk files:
- event_tracer.cc
* Android relevant upstream changes:
- Make Beamforming dynamically settable for Android platform builds
- Remove additional channel constraints when Beamforming is enabled in AudioProcessing
- Use an explicit identifier in Config
Change-Id: I384a4e8f6982c31c5bc70eef521bb2d90800b9a4
|
|
BUG=webrtc:5263
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1589523002 .
Cr-Commit-Position: refs/heads/master@{#11255}
|
|
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}
|
|
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}
|
|
TBR=pthatcher@webrtc.org
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1588083002 .
Cr-Commit-Position: refs/heads/master@{#11252}
|
|
Chromium reported errors when building libjingle_nacl due to some methods used virtual instead of override when they were overriding the base class. My guess is that when one method starts using override, all other in the same class must too.
R=tommi@webrtc.org
TBR=pthatcher@webtrc.org
BUG=4173
Review URL: https://codereview.webrtc.org/1589563003 .
Cr-Commit-Position: refs/heads/master@{#11251}
|
|
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}
|
|
AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
Reason for revert:
Broke Chrome:
https://build.chromium.org/p/tryserver.chromium.linux/builders/linux_chromium_chromeos_compile_dbg_ng/builds/143025/steps/compile%20%28with%20patch%29/logs/stdio
FAILED: cd ../../third_party/libjingle; python ../../native_client/build/build_nexe.py --root ../.. --product-dir ../../out/Debug/xyz --config-name Debug -t ../../native_client/toolchain/ --arch pnacl --build newlib_plib --name ../../out/Debug/gen/tc_pnacl_newlib/lib/libjingle_nacl.a --objdir ../../out/Debug/obj/third_party/libjingle/libjingle_nacl.gen/pnacl_newlib-pnacl/libjingle_nacl "--include-dirs=../../out/Debug/gen/tc_pnacl_newlib/include ../.. \"../../out/Debug/gen\" ./source ../ ../../native_client_sdk/src/libraries ../../native_client_sdk/src/libraries/nacl_io/include ../../native_client_sdk/src/libraries/third_party/newlib-extras ../expat/files/lib ../boringssl/src/include" "--compile_flags=-O2 -g -Wall -fdiagnostics-show-option -Werror -Wno-unused-function -Wno-char-subscripts -Wno-c++11-extensions -Wno-unnamed-type-template-args -Wno-extra-semi -Wno-unused-private-field -Wno-char-subscripts -Wno-unused-function \"-std=gnu++11\" " --gomadir /b/build/goma "--defines=\"__STDC_LIMIT_MACROS=1\" \"__STDC_FORMAT_MACROS=1\" \"_GNU_SOURCE=1\" \"_POSIX_C_SOURCE=199506\" \"_XOPEN_SOURCE=600\" \"DYNAMIC_ANNOTATIONS_ENABLED=1\" \"DYNAMIC_ANNOTATIONS_PREFIX=NACL_\" \"NACL_BUILD_ARCH=x86\" V8_DEPRECATION_WARNINGS \"CLD_VERSION=2\" \"_FILE_OFFSET_BITS=64\" CHROMIUM_BUILD \"CR_CLANG_REVISION=255169-1\" COMPONENT_BUILD UI_COMPOSITOR_IMAGE_TRANSPORT \"USE_AURA=1\" \"USE_ASH=1\" \"USE_PANGO=1\" \"USE_CAIRO=1\" \"USE_DEFAULT_RENDER_THEME=1\" \"USE_LIBJPEG_TURBO=1\" \"USE_X11=1\" \"IMAGE_LOADER_EXTENSION=1\" \"ENABLE_WEBRTC=1\" \"ENABLE_MEDIA_ROUTER=1\" USE_PROPRIETARY_CODECS ENABLE_PEPPER_CDMS ENABLE_CONFIGURATION_POLICY ENABLE_NOTIFICATIONS \"ENABLE_HIDPI=1\" \"ENABLE_TOPCHROME_MD=1\" USE_UDEV DONT_EMBED_BUILD_METADATA \"DCHECK_ALWAYS_ON=1\" FIELDTRIAL_TESTING_ENABLED \"ENABLE_TASK_MANAGER=1\" \"ENABLE_EXTENSIONS=1\" \"ENABLE_PDF=1\" \"ENABLE_PLUGINS=1\" \"ENABLE_SESSION_SERVICE=1\" \"ENABLE_THEMES=1\" \"ENABLE_AUTOFILL_DIALOG=1\" \"ENABLE_BACKGROUND=1\" \"ENABLE_PRINTING=1\" \"ENABLE_PRINT_PREVIEW=1\" \"ENABLE_SPELLCHECK=1\" \"ENABLE_CAPTIVE_PORTAL_DETECTION=1\" \"ENABLE_APP_LIST=1\" \"ENABLE_SUPERVISED_USERS=1\" \"ENABLE_MDNS=1\" \"ENABLE_SERVICE_DISCOVERY=1\" V8_USE_EXTERNAL_STARTUP_DATA FULL_SAFE_BROWSING SAFE_BROWSING_CSD SAFE_BROWSING_DB_LOCAL EXPAT_RELATIVE_PATH FEATURE_ENABLE_SSL GTEST_RELATIVE_PATH HAVE_OPENSSL_SSL_H NO_MAIN_THREAD_WRAPPING NO_SOUND_SYSTEM WEBRTC_POSIX SRTP_RELATIVE_PATH SSL_USE_OPENSSL USE_WEBRTC_DEV_BRANCH \"timezone=_timezone\" XML_STATIC \"USE_LIBPCI=1\" \"USE_OPENSSL=1\" \"USE_OPENSSL_CERTS=1\"" "--link_flags=-B../../out/Debug/gen/tc_pnacl_newlib/lib " "--source-list=../../out/gypfiles/third_party/libjingle/pnacl_newlib.libjingle_nacl.source_list.gypcmd"
In file included from ../webrtc/p2p/base/tcpport.cc:67:
../webrtc/p2p/base/tcpport.h:50:23: error: 'CreateConnection' overrides a member function but is not marked 'override' [-Werror,-Winconsistent-missing-override]
virtual Connection* CreateConnection(const Candidate& address,
^
../webrtc/p2p/base/portinterface.h:71:23: note: overridden virtual function is here
virtual Connection* CreateConnection(
^
In file included from ../webrtc/p2p/base/tcpport.cc:67:
../webrtc/p2p/base/tcpport.h:53:16: error: 'PrepareAddress' overrides a member function but is not marked 'override' [-Werror,-Winconsistent-missing-override]
virtual void PrepareAddress();
^
../webrtc/p2p/base/portinterface.h:63:16: note: overridden virtual function is here
virtual void PrepareAddress() = 0;
^
(etc)
Original issue's description:
> Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
>
> To reduce the risk of future mistakes when connecting Ports, Port::OnSentPacket was made pure virtual to ensure that new implementations take care of it.
>
> BUG=4173
> R=pthatcher@webrtc.org
>
> Committed: https://crrev.com/7307952a5bf63311e5f9a2a90089a06177e42716
> Cr-Commit-Position: refs/heads/master@{#11247}
TBR=pthatcher@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=4173
Review URL: https://codereview.webrtc.org/1586063002
Cr-Commit-Position: refs/heads/master@{#11249}
|
|
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}
|
|
To reduce the risk of future mistakes when connecting Ports, Port::OnSentPacket was made pure virtual to ensure that new implementations take care of it.
BUG=4173
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1577873003 .
Cr-Commit-Position: refs/heads/master@{#11247}
|
|
Also add a perf metric tracking the average network latency.
The audio stream test is disabled for now since audio isn't included in bitrate allocation.
BUG=webrtc:5263
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1582833002 .
Cr-Commit-Position: refs/heads/master@{#11244}
|
|
Previosly ToSesnsetiveString() wasn't working witn some implementations
of inet_ntop(). Rewrote it to avoid that dependency.
BUG=chromium:577344
R=pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1584793004 .
Cr-Commit-Position: refs/heads/master@{#11242}
|
|
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}
|
|
Fixed in latest libvpx roll.
Keep EndToEndTest.TransportSeqNumOnAudioAndVideo disabled on
Win_DrMemory for now as it seems to time-out/too slow.
TBR=stefan@webrtc.org, kjellander@webrtc.org
BUG=webrtc:5402
NOTRY=True
Review URL: https://codereview.webrtc.org/1577313003
Cr-Commit-Position: refs/heads/master@{#11240}
|
|
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}
|
|
This makes it possible to use protobuffers with
an external protobuf library instead of the one that
comes with the WebRTC code.
NOTRY=True
Review URL: https://codereview.webrtc.org/1589433002
Cr-Commit-Position: refs/heads/master@{#11236}
|
|
BUG=webrtc:5255
Review URL: https://codereview.webrtc.org/1578373003
Cr-Commit-Position: refs/heads/master@{#11235}
|
|
the way
Class renamed to indicated use of the rtcp::Empty class: it can only be used as container for other rtcp packets.
All tests that use Append function moved from rtcp_packet_unittest, even if they did not use Empty class.
This is because there is plan to make RtcpPacket class lighter by moving Append functionality to CompoundPacket class.
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1582503002
Cr-Commit-Position: refs/heads/master@{#11234}
|
|
I think the problem was that I only introduced delay in one direction, and the estimation assumes that the RTT is evenly divided between the send direction and the receive direction, which was true for the old test.
BUG=chromium:576246
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1577853005 .
Cr-Commit-Position: refs/heads/master@{#11233}
|
|
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}
|
|
BUG=webrtc:5250
Review URL: https://codereview.webrtc.org/1551813002
Cr-Commit-Position: refs/heads/master@{#11230}
|
|
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}
|
|
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1551893002
Cr-Commit-Position: refs/heads/master@{#11228}
|
|
It looks to me like targets :rtc_base_approved is logically a subset of
:rtc_base, and so any targets depending on :rtc_base expect to also get
access to the headers in :rtc_base_approved.
Thus I think it's appropriate for :rtc_base to have :rtc_base_approved in
public_deps, so that `gn check` will permit this without clients having to
explicitly depend on both.
NOTRY=True
Review URL: https://codereview.webrtc.org/1578833002
Cr-Commit-Position: refs/heads/master@{#11227}
|
|
TBR=minyue@webrtc.org
NOTRY=true
Review URL: https://codereview.webrtc.org/1578223003 .
Cr-Commit-Position: refs/heads/master@{#11226}
|
|
There's no reason not to use std::move instead now that we can use the
C++11 standard library.
BUG=webrtc:5373
Review URL: https://codereview.webrtc.org/1531013003
Cr-Commit-Position: refs/heads/master@{#11225}
|
|
BUG=chromium:572995
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1578223002 .
Cr-Commit-Position: refs/heads/master@{#11224}
|
|
BUG=
Review URL: https://codereview.webrtc.org/1582523002
Cr-Commit-Position: refs/heads/master@{#11221}
|
|
Also wires up receiving transport sequence numbers.
BUG=webrtc:5263
R=mflodman@webrtc.org, pbos@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1535963002 .
Cr-Commit-Position: refs/heads/master@{#11220}
|
|
The unit test will run the pure C denoiser and SSE2/NEON denoiser (based
on the CPU detection) and compare the denoised frames to ensure the bit
exact.
TBR=tommi@webrtc.org
BUG=webrtc:5255
Review URL: https://codereview.webrtc.org/1492053003
Cr-Commit-Position: refs/heads/master@{#11216}
|
|
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}
|
|
Review URL: https://codereview.webrtc.org/1563493005
Cr-Commit-Position: refs/heads/master@{#11213}
|
