The AEC dump was not self-contented enough in the sense that APM configuration is missing, and therefore, given an AEC dump, it is sometimes not clear how to reproduce problems.
This CL tries to address the problem.
Note that this cannot guarantee a perfect reproduction in all cases. Dumping from the middle of a call makes the initial states unknown and thus may make the result non-reproducible.
BUG=
TEST= 1. new dump in Chromium and unpack
2. unpack old dump
R=andrew@webrtc.org, peah@webrtc.org
Review URL: https://codereview.webrtc.org/1348903004 .
Cr-Commit-Position: refs/heads/master@{#10155}
Microsoft introduced modern app from win8. Modern apps can be used cross Microsoft's platforms.
It was confirmed from Microsoft that there is no support for modern app's window capture.
BUG=526883
Review URL: https://codereview.webrtc.org/1371383003
Cr-Commit-Position: refs/heads/master@{#10154}
Adds a loopback button that will connect to itself by simulating another client connection to the web socket server.
Adds an audio only mode switch.
BUG=
Review URL: https://codereview.webrtc.org/1334003002
Cr-Commit-Position: refs/heads/master@{#10153}
Poller thread is currently started in the constructor, so the first call
to PollStats() may happen even before the streams have been configured.
This will blow up on RTC_DCHECK_GT(expected_bitrate_bps_, 0);
Thread should instead be started on PerformTest() call.
BUG=
Review URL: https://codereview.webrtc.org/1378303004
Cr-Commit-Position: refs/heads/master@{#10149}
This allows us to pass packet meta data, such as transport sequence
number, to libjingle and further down to the socket implementation. A
similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h.
BUG=4173
Review URL: https://codereview.webrtc.org/1376673004
Cr-Commit-Position: refs/heads/master@{#10144}
This CL changes the threshold where we consider a block to be static and
of sufficient quality to not spend bits/CPU encoding it.
Perf note: This change may result in a minor degradation of PSNR/SSIM
and available send bitrate. CPU usage and bitrate sent should however
be greately reduced.
BUG=webrtc:5015
Review URL: https://codereview.webrtc.org/1383533002
Cr-Commit-Position: refs/heads/master@{#10134}
In particular, if 14 short deltas were inserted (2 * capacity of status
vector chunk with 2bit items) followed by a large delta, that status
item would be dropped.
BUG=
Review URL: https://codereview.webrtc.org/1367193002
Cr-Commit-Position: refs/heads/master@{#10132}
We used to link with all audio codecs unconditionally (except Opus);
this patch makes gyp and gn only link to the ones that are used.
This unfortunately fails to have a measurable impact on Chromium
binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
fix were already being excluded from Chromium by some other means,
likely just the linker omitting compilation units with no incoming
references.
(This was previously landed as revisions 10046 and 10060, and got
reverted because it broke several of the Chromium FYI bots.)
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1368843003
Cr-Commit-Position: refs/heads/master@{#10127}
Reason for revert:
This broke chromium.fyi bot.
Original issue's description:
> Change WebRTC SslCipher to be exposed as number only.
>
> This makes the SSL exposed as uint16_t which is the IANA value. GetRfcSslCipherName is introduced to handle the conversion to names from ID. IANA value will be used for UMA reporting. Names will still be used for WebRTC stats reporting.
>
> For SRTP, currently it's still string internally but is reported as IANA number.
>
> This is used by the ongoing CL https://codereview.chromium.org/1335023002.
>
> BUG=523033
>
> Committed: https://crrev.com/4fe3c9a77386598db9abd1f0d6983aefee9cc943
> Cr-Commit-Position: refs/heads/master@{#10124}
TBR=juberti@webrtc.org,rsleevi@chromium.org,pthatcher@webrtc.org,davidben@chromium.org,juberti@google.com,davidben@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=523033
Review URL: https://codereview.webrtc.org/1380603005
Cr-Commit-Position: refs/heads/master@{#10125}
This makes the SSL exposed as uint16_t which is the IANA value. GetRfcSslCipherName is introduced to handle the conversion to names from ID. IANA value will be used for UMA reporting. Names will still be used for WebRTC stats reporting.
For SRTP, currently it's still string internally but is reported as IANA number.
This is used by the ongoing CL https://codereview.chromium.org/1335023002.
BUG=523033
Review URL: https://codereview.webrtc.org/1337673002
Cr-Commit-Position: refs/heads/master@{#10124}
This change filters out local ports when CF_HOST is not originally specified to prevent these ports from sending out STUN which leaks IP address.
BUG=webrtc:4946
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1378753003 .
Cr-Commit-Position: refs/heads/master@{#10121}
Connecting TransportChannelImpls directly to the TransportController,
and removing redundant signal forwarding/state aggregating code from
Transport. This brings us closer to just getting rid of Transport
entirely.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1380563002 .
Cr-Commit-Position: refs/heads/master@{#10120}
Our perf test suite webrtc_perf_tests timed out, which caused most
of the delay landing this (https://crbug.comn/535973 and
https://codereview.chromium.org/1370133004).
Other problems with executing Android tests also needed to be
resolved in order to land this (http://crbug.com/534849).
Libvpx has moved from third_party/libvpx to third_party/libvpx_new
as of https://codereview.chromium.org/1323333002/
Android GN was blocking this roll due to a problem that ended up
being caused by a bug (http://crbug.com/534849).
Relevant changes:
* src/buildtools: f7310ee..8d89c1b
* src/third_party/boringssl/src: 1d128f3..4c60d35
* src/third_party/icu: 6b3ce81..423fc7e
* src/third_party/libjpeg_turbo: 631e2dd..e4e7503
* src/third_party/libvpx: ac1772e..70db223
* src/third_party/libyuv: fcacbfb..62c49dc
* src/tools/gyp: 5d01a8c..01528c7
* src/tools/swarming_client: 77f720b..6e5d2b2
Details: 310ea93..8cf53d6/DEPS
Clang version changed 245965:247874
Details: 310ea93..8cf53d6/tools/clang/scripts/update.sh
BUG=481034, 535973
TBR=marpan@webrtc.org
Review URL: https://codereview.webrtc.org/1355083002
Cr-Commit-Position: refs/heads/master@{#10101}
Fixes code formatting and uses size_t properly. Also makes use of
IsNewerTimestamp instead of a simple > check, which should fix an
edge-case bug.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1358863002
Cr-Commit-Position: refs/heads/master@{#10094}
whose network has ever been removed. It is unlikely the sockets/ports/candidates created from
those AllocationSequences will still be valid.
BUG=
Review URL: https://codereview.webrtc.org/1361183004
Cr-Commit-Position: refs/heads/master@{#10093}
p2ptransportchannel. This CL does not use the new policy yet.
BUG=
Review URL: https://codereview.webrtc.org/1369773003
Cr-Commit-Position: refs/heads/master@{#10092}
The playout mode in NetEq can still be set through the constructor
configuration.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1362943004
Cr-Commit-Position: refs/heads/master@{#10089}
In addition to this the ramp-up tests are refactored to use a receive call instead of only a remote bitrate estimator, and to make use of BaseTest.
BUG=webrtc:4836
Review URL: https://codereview.webrtc.org/1368943002
Cr-Commit-Position: refs/heads/master@{#10087}
This updates the isolate.gypi copies we have to maintain in our
code repo to Chromium's revision 310ea93.
The changes about generating .isolated.gen.json files are needed
to support running with Swarming (https://www.chromium.org/developers/testing/isolated-testing)
Since isolated testing is now using a new launch script
in tools: isolate_driver.py, that's added to our links
script.
In order to use isolate_driver.py, the .isolate files must be in the
same directory as the test_name_run target is defined, which meant
I had to move around some of the isolate files and targets below
webrtc/modules.
BUG=497757
R=maruel@chromium.orgTBR=henrik.lundin@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org
TESTED=Clobbered trybots:
git cl try -c --bot=linux_compile_rel --bot=mac_compile_rel --bot=win_compile_rel --bot=android_compile_rel --bot=ios_rel -m tryserver.webrtc
Review URL: https://codereview.webrtc.org/1373513002 .
Cr-Commit-Position: refs/heads/master@{#10081}
It has become extra flaky lately, and is preventing people from
using the CQ.
BUG=webrtc:4958
Review URL: https://codereview.webrtc.org/1368763002
Cr-Commit-Position: refs/heads/master@{#10080}
Android hardware H.264 seems to keep a steady high-QP flow instead of
dropping frames, so framedrops aren't sufficient to detect a bad state
where downscaling would be beneficial.
BUG=webrtc:4968
R=magjed@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1364253002 .
Cr-Commit-Position: refs/heads/master@{#10078}
