Currently,
https://build.chromium.org/p/chromium.fyi/builders/CrWinAsan%28dll%29/builds/...
is broken because of
Writing """\
clang_use_chrome_plugins = false
is_asan = true
is_clang = true
is_component_build = true
is_debug = false
llvm_force_head_revision = true
symbol_level = 2
target_cpu = "x86"
v8_enable_verify_heap = true
""" to C:\b\c\b\CrWinAsan_dll_\src\out\Release\args.gn.
C:\b\c\b\CrWinAsan_dll_\src\buildtools\win\gn.exe gen //out/Release --check
--runtime-deps-list-file=C:\b\c\b\CrWinAsan_dll_\src\out\Release\runtime_deps
-> returned 1
ERROR at //third_party/webrtc/base/BUILD.gn:276:21: Undefined identifier in string expansion.
data += [ "$clang_base_path/lib/clang/$clang_version/lib/windows/clang_rt.asan_dynamic-i386.dll" ]
^--------------
"clang_base_path" is not currently in scope.
See //content/test/BUILD.gn:307:7: which caused the file to be included.
"//third_party/webrtc/base:rtc_base",
^-----------------
NOTRY=True
NOPRESUBMIT=True
R=kjellander@webrtc.org, thakis@chromium.org
BUG=chromium:497757
TBR=kjellander
Review-Url: https://codereview.webrtc.org/2422223002
Cr-Commit-Position: refs/heads/master@{#14653}
Reason for revert:
Speculative revert as it may be the cause of the DrMemory test failure:
https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/builds/5115
Original issue's description:
> Add path for recovered packets from internal::Call to RtpStreamReceiver.
>
> When the FlexfecReceiver recovers media packets, it inserts these into
> internal::Call, which then distributes them to the appropriate
> VideoReceiveStream/RtpStreamReceiver.
>
> BUG=webrtc:5654
>
> Committed: https://crrev.com/9c4b4b47f4325b48e1856566a30983f9e4e30dd0
> Cr-Commit-Position: refs/heads/master@{#14642}
TBR=stefan@webrtc.org,brandtr@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2427733002
Cr-Commit-Position: refs/heads/master@{#14652}
This CL enables printing Android log to stdout for the test. This makes debugging the flakiness of the test easier.
BUG=webrtc:6475
R=magjed@webrtc.org
Review-Url: https://codereview.webrtc.org/2418413002
Cr-Commit-Position: refs/heads/master@{#14649}
for consistency with other rtcp packet classes.
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2361853002
Cr-Commit-Position: refs/heads/master@{#14648}
and thus IP_PACKET_SIZE constant:
Build() use BlockLength() instead of constant IP_PACKET_SIZE for packet
capacity, adding extra checks about packet generation in tests.
Build(callback) removed as unused.
definitions reordered to follow style.
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2270753002
Cr-Commit-Position: refs/heads/master@{#14647}
When the FlexfecReceiver recovers media packets, it inserts these into
internal::Call, which then distributes them to the appropriate
VideoReceiveStream/RtpStreamReceiver.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2390823009
Cr-Commit-Position: refs/heads/master@{#14642}
AEC and AECM when these become full to also work when not
in debug mode.
BUG=webrtc:6530
Review-Url: https://codereview.webrtc.org/2419023002
Cr-Commit-Position: refs/heads/master@{#14637}
To ensure this change won't break Chromium, this is the first change, to add a
new CaptureFrame() function, and let Capture(DesktopRegion) and CaptureFrame()
call each other. So both a legacy consumer or a legacy implementation won't be
broken.
BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=6513
Review-Url: https://codereview.webrtc.org/2409833002
Cr-Commit-Position: refs/heads/master@{#14635}
I added a few missing dependencies to the GN target of voice_engine while doing other
unrelated work. Currently GN's header include checker has the
following to say:
$ gn check out/gn_debug webrtc/voice_engine
ERROR at //webrtc/voice_engine/include/voe_network.h:38:11: Include not allowed.
#include "webrtc/transport.h"
^-----------------
It is not in any dependency of
//webrtc/voice_engine:voice_engine
The include file is in the target(s):
//webrtc:webrtc
which should somehow be reachable.
transport.h should probably move in to webrtc/api, since it is already
a pure virtual interface and is used in quite a few places.
BUG=webrtc:5589
NOTRY=True
Review-Url: https://codereview.webrtc.org/2421483002
Cr-Commit-Position: refs/heads/master@{#14633}
asyncstuntcpsocket_unittest.cc never seem to have been added
along with the other tests in webrtc/p2p. Luckily the tests pass.
BUG=None
NOTRY=True
Review-Url: https://codereview.webrtc.org/2404173002
Cr-Commit-Position: refs/heads/master@{#14632}
Reason for revert:
Speculative revert.
Intermittent memory access errors suspected to be caused by this cl.
See for instance https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/8018
UNADDRESSABLE ACCESS of freed memory: reading 0x0331d330-0x0331d334 4 byte(s)
# 0 webrtc::voe::RtcpRttStatsProxy::LastProcessedRtt
# 1 webrtc::ModuleRtpRtcpImpl::Process
Original issue's description:
> Add RtcpRttStats to AudioStream
>
> BUG=webrtc:6508
>
> Committed: https://crrev.com/e0729c56d35acfaf9738fdb32c6508cd78eaf089
> Cr-Commit-Position: refs/heads/master@{#14595}
TBR=stefan@webrtc.org,minyue@webrtc.org,solenberg@webrtc.org,michaelt@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6508
Review-Url: https://codereview.webrtc.org/2415943002
Cr-Commit-Position: refs/heads/master@{#14631}
With this change, the calculations inside AverageIAT are changed to be
in double-precision floating point instead of in fixed point. Also,
the method's name is changed to EstimatedClockDriftPpm to better
reflect what it returns.
A few unit tests had to be updated because of minor numerical
differences.
Also removing the UBSan suppression related to this issue.
BUG=webrtc:5889
Review-Url: https://codereview.webrtc.org/2408653002
Cr-Commit-Position: refs/heads/master@{#14628}
This was an ill tested special case which turned out to be more problem
than benefit. The special case was only triggered when the decoder frame
size was smaller than 10 ms, which is more or less unsupported by NetEq.
Also fixed a bug in a test, a bug which was exposed by the code change.
BUG=chromium:654983
Review-Url: https://codereview.webrtc.org/2412883002
Cr-Commit-Position: refs/heads/master@{#14627}
Reason for revert:
There is a risk of ending up in a bad state due to race conditions with this patch. Tests in downstream clients have shown that it can
happen that an output stream is opened up in MUSIC mode when it should not.
Reverting since the new functionality added here is not worth the
risk of breaking existing clients.
Original issue's description:
> Android audio playout now supports non-call media streams.
>
> The default (preferred) stream type for output audio is STREAM_VOICE_CALL since the WebRTC stack is mainly intended for VoIP calls. But if the user wants to run in another mode than COMM mode, we now accept it and change the stream type to STREAM_MUSIC instead. It can e.g. be suitable for applications that does not record audio or if a call shall be casted to a Chromecast device.
>
> The solution is somewhat experimental.
>
> NOTRY=TRUE
>
> BUG=webrtc:4767
>
> Committed: https://crrev.com/872f614111f436d15e29516ce19c3b63d25b8639
> Cr-Commit-Position: refs/heads/master@{#14613}
TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4767
Review-Url: https://codereview.webrtc.org/2420583002
Cr-Commit-Position: refs/heads/master@{#14626}
Reason for revert:
Seems to break our Android Swarming bots. Probably due to https://cs.chromium.org/chromium/src/testing/buildbot/gn_isolate_map.pyl?rcl=0&l=875
Error:
/b/c/b/Android32__M_Nexus5X__dbg_/src/buildtools/linux64/gn gen //out/Debug --check --runtime-deps-list-file=/b/c/b/Android32__M_Nexus5X__dbg_/src/out/Debug/runtime_deps
-> returned 1
ERROR The label "//webrtc/tools:tools_unittests(//build/toolchain/android:arm)" isn't a target.
When reading the line:
//webrtc/tools:tools_unittests
from the --runtime-deps-list-file=/b/c/b/Android32__M_Nexus5X__dbg_/src/out/Debug/runtime_deps
GN gen failed: 1
Original issue's description:
> Remove tools dir from root webrtc target
>
> Removing it as we don't need it to build as part of webrtc target.
>
> BUG=webrtc:6412
> NOTRY=True
>
> Committed: https://crrev.com/163b1a2d0a0f8e822d8cd15f6385057bc7988ad1
> Cr-Commit-Position: refs/heads/master@{#14622}
TBR=charujain@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6412
Review-Url: https://codereview.webrtc.org/2420573002
Cr-Commit-Position: refs/heads/master@{#14624}
Removing it as we don't need it to build as part of webrtc target.
BUG=webrtc:6412
NOTRY=True
Review-Url: https://codereview.webrtc.org/2412353004
Cr-Commit-Position: refs/heads/master@{#14622}
The default implementations are provided as to not break Chromium mocks,
as soon as we have done a successful roll they should be updated and the
default implementations removed.
TBR=hta@webrtc.org, deadbeef@webrtc.org
NOTRY=True
BUG=chromium:654927
Review-Url: https://codereview.webrtc.org/2414613003
Cr-Commit-Position: refs/heads/master@{#14617}
Make sure that the appropriate run loop source gets added/removed. More clean up
to remove unnecessary functions and suppress deprecated declaration warnings.
BUG=webrtc:6029
Review-Url: https://codereview.webrtc.org/2417603002
Cr-Commit-Position: refs/heads/master@{#14615}
This gets rid of a bit of codec-specific code in VoE.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2355483003
Cr-Commit-Position: refs/heads/master@{#14614}
The default (preferred) stream type for output audio is STREAM_VOICE_CALL since the WebRTC stack is mainly intended for VoIP calls. But if the user wants to run in another mode than COMM mode, we now accept it and change the stream type to STREAM_MUSIC instead. It can e.g. be suitable for applications that does not record audio or if a call shall be casted to a Chromecast device.
The solution is somewhat experimental.
NOTRY=TRUE
BUG=webrtc:4767
Review-Url: https://codereview.webrtc.org/2411263003
Cr-Commit-Position: refs/heads/master@{#14613}
Changed mixability status into AddSource/RemoveSource. Added 'ssrc()'
method to the MixerSource interface. Removed unnecessary member 'num_audio_sources_' and made the mixer be refcounted.
BUG=webrtc:6346
NOTRY=True
Review-Url: https://codereview.webrtc.org/2408683002
Cr-Commit-Position: refs/heads/master@{#14612}
to the functionality in the audio processing module.
Therefore, it should be a pure interface.
This CL ensures that is the case.
BUG=webrtc:6515
Review-Url: https://codereview.webrtc.org/2406193002
Cr-Commit-Position: refs/heads/master@{#14608}
Reason for revert:
Breaks upstream code.
Original issue's description:
> Prune connections based on network name.
> Previously we prune connections on the same network pointer.
> So if an IPv6 and an IPv4 network are on the same network interface, IPv4 connection won't be pruned even if an IPv6 connection with higher priority becomes writable.
>
> With this change, as long as one connection becomes writable, all connections having lower priority with the same network name will be pruned.
>
> Also simplify the implementation.
>
> BUG=webrtc:6512
>
> Committed: https://crrev.com/aae2784c1fab9d1510393dec15d76caa574e2da8
> Cr-Commit-Position: refs/heads/master@{#14593}
TBR=skvlad@webrtc.org,honghaiz@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6512
Review-Url: https://codereview.webrtc.org/2412433003
Cr-Commit-Position: refs/heads/master@{#14601}
MixerAudioSource is moved to AudioMixerImpl::Source. Structures and methods of the MixerAudioSource interface have been renamed. The RemixFrame method has added checks and is moved to audio_frame_manipulator.h
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2396803004
Cr-Commit-Position: refs/heads/master@{#14600}
After calling Start(), doing a Stop() then Start() sequence should bring
the stream back to the original state.
BUG=webrtc:6501
Review-Url: https://codereview.webrtc.org/2407163002
Cr-Commit-Position: refs/heads/master@{#14597}
