If the local port and the remote candidate's protocols do not match,
do not even try to pair them.
This avoids printing out confusing logs like
"Attempt to change a remote candidate..." in p2ptransportchannel
when two remote candidates have the same port number but different
protocols.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1516613002 .
Cr-Commit-Position: refs/heads/master@{#11034}
Removes noisy dependencies on webrtc_fuzzer_main and removal of
find_bad_constructs, removes 1-6 lines of gn per fuzzer target.
BUG=webrtc:4771
R=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1524993002 .
Cr-Commit-Position: refs/heads/master@{#11022}
rand() usage replaced with new Random class, which also makes it clearer what interval random number is in.
BUG=webrtc:5277
R=mflodman
Review URL: https://codereview.webrtc.org/1519503002
Cr-Commit-Position: refs/heads/master@{#11019}
except rand() function that is subject of CL#1519503002
and namespace that is fixed in CL#1506823002
BUG=webrtc:5277
R=mflodman
Review URL: https://codereview.webrtc.org/1511413005
Cr-Commit-Position: refs/heads/master@{#11012}
The input resolution (output from video_adapter) can be further scaled down or higher video layer(s) can be dropped due to bitrate constraints.
BUG=webrtc:4112
Review URL: https://codereview.webrtc.org/1502173002
Cr-Commit-Position: refs/heads/master@{#11006}
This makes it clearer that the IP address has been stripped.
BUG=chromium:254329
Review URL: https://codereview.webrtc.org/1516163003
Cr-Commit-Position: refs/heads/master@{#11005}
AudioProcessing::ProcessReverseStream().
Before the change, only -1 was considered to be an error.
Allthough the error code scheme for AudioProcessing definitely
could be discussed, the current scheme have many error codes that
differ from -1 and thus were not caught by the old code.
BUG=webrtc:5237
Review URL: https://codereview.webrtc.org/1515073004
Cr-Commit-Position: refs/heads/master@{#11003}
We already had a special case for android, but it only worked for arm32.
BUG=webrtc:4198, webrtc:4199
Review URL: https://codereview.webrtc.org/1512833003
Cr-Commit-Position: refs/heads/master@{#10989}
As a step toward fixing webrtc:3987, here we update the RTPencode to allow Opus RTP payloads.
BUG=webrtc:3987, webrtc:2692
Review URL: https://codereview.webrtc.org/1516653003
Cr-Commit-Position: refs/heads/master@{#10987}
rtc::PlatformThreadId is pid_t (32-bit signed int) on Linux and Mac,
but DWORD (32-bit unsigned int) on Windows.
Using the %d printf specifier is therefore not correct on Windows,
and Clang would warn about it:
..\..\third_party\webrtc\base\event_tracer.cc(124,46) : error: format specifies
type 'int' but the argument has type 'rtc::PlatformThreadId' (aka 'unsigned
long') [-Werror,-Wformat]
e.phase, e.timestamp, e.pid, e.tid);
^~~~~
This commit fixes the problem by explicitly casting to int before printing.
BUG=82385
Review URL: https://codereview.webrtc.org/1514253002 .
Cr-Commit-Position: refs/heads/master@{#10982}
Fixes one sign mismatch warning, and one "const has no effect and is
ignored" warning.
BUG=chromium:567877
Review URL: https://codereview.webrtc.org/1510233002
Cr-Commit-Position: refs/heads/master@{#10976}
The purpose is so that a decoder (Android) that only have a limited number of output buffers can make sure that decoding is done just before the frame is needed.
Removed unused iSupportsRenderTiming and the settings structs since it was not used.
Added VCMReceiver::FrameForDecoding unit test for the case when PreferDecodeLate is set.
Note that this does not change the current behaviour. We actually currently always decode frames late. This cl is to make sure the behaviour is kept for Android, if the default behaviour is changed.
Review URL: https://codereview.webrtc.org/1428293003
Cr-Commit-Position: refs/heads/master@{#10974}
By reducing the length of the audio input, the total runtime of
$ out/Debug/modules_tests --gtest_filter=AudioCodingModuleTest.*
is reduced by more than 10x, when run single-threaded.
The PCMFile helper class is extended with a FastForward method (to
skip initial silence in the test files) and a limiter on how much to
read.
BUG=webrtc:2463
R=ivoc@webrtc.org
Review URL: https://codereview.webrtc.org/1513223002 .
Cr-Commit-Position: refs/heads/master@{#10973}
They were meant to be run if we have either iSAC float or fix, but the
typo made them run for just float.
BUG=webrtc:4198, webrtc:4199
Review URL: https://codereview.webrtc.org/1513483005
Cr-Commit-Position: refs/heads/master@{#10969}
Makes use of rtc::Event which is simpler and can be used without
allocating additional objects on the heap.
Does not modify test/channel_transport/.
BUG=
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1487893004 .
Cr-Commit-Position: refs/heads/master@{#10968}
rtcp_utility, rtp_utility, tmmbr_help, rtcp_receiver, rtcp_receiver_help are explicetly excluded from the cleanup becaues there are short plans (or cls) to do a deeper cleaning there.
BUG=webrtc:5277
R=pbos@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1512493002
Cr-Commit-Position: refs/heads/master@{#10966}
Created a simple unit test for the new random number generator. (It mostly tests
that the generated numbers are consistent with the intended distribution, e.g. uniform.
It is not a comprehensive test of the quality of the random numbers.)
Several assertions in OveruseDetectorTest seem to depend on the exact sequence of random numbers. I updated those numbers to work with the new PRNG.
Compute the standard deviation of the expected result in TestReorderFilter instead of passing an uncertainty parameter.
BUG=webrtc:5177
Review URL: https://codereview.webrtc.org/1457023002
Cr-Commit-Position: refs/heads/master@{#10965}
This CL makes sure no RTCP SR is sent before there is a valid timestamp
to set in the SR, based on the first sent media packet.
BUG=webrtc:1600
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1506103006 .
Cr-Commit-Position: refs/heads/master@{#10964}
