The paths of the protobuf output files needs to match the actual tree path
in order for Bazel builds to work.
BUG=webrtc:6412
NOTRY=True
Review-Url: https://codereview.webrtc.org/2716413004
Cr-Commit-Position: refs/heads/master@{#16912}
This is a follow-up to https://codereview.webrtc.org/2670643007/. That
CL provided significant improvement to Mac, Linux and ARM-based
platforms, but failed to improve the performance for Windows. The
problem is that the MSVC compiler did not produce branch-free code for
that fix. This new change produces the same result for non-Windows
platforms, as well as introduces branch-free code for Windows.
H/t to kwiberg@ for providing the solution.
BUG=webrtc:7159
Review-Url: https://codereview.webrtc.org/2700633003
Cr-Commit-Position: refs/heads/master@{#16649}
After https://codereview.webrtc.org/2340773002,
the path from webrtc::test::ResourcePath in
webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc is wrong.
It is
/path/to/repos/resources/audio_coding/testfile32kHz.pcm
It should be
/path/to/repos/webrtc-temp/src/resources/audio_coding/testfile32kHz.pcm.
The middle part is missing.
The reason this target is affected is because
webrtc::test::SetExecutablePath(argv[0]);
was not called.
That call is necessary for us to know that the test is being run from src/
and not from out/Default (as is assumed, when that function is not called.)
BUG=chromium:497757
R=kjellander@webrtc.org, henrik.lundin@webrtc.org
Review-Url: https://codereview.webrtc.org/2698743002
Cr-Commit-Position: refs/heads/master@{#16641}
As it is, the test fails to compile on some downstream compilers with the following error:
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc:316:25: error: lambda capture 'kPayloadLength' is not required to be captured for this use [-Werror,-Wunused-lambda-capture]
.WillOnce(Invoke([kPayloadLength, kFirstSequenceNumber, kFirstTimestamp,
^
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc:316:41: error: lambda capture 'kFirstSequenceNumber' is not required to be captured for this use [-Werror,-Wunused-lambda-capture]
.WillOnce(Invoke([kPayloadLength, kFirstSequenceNumber, kFirstTimestamp,
^
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc:316:63: error: lambda capture 'kFirstTimestamp' is not required to be captured for this use [-Werror,-Wunused-lambda-capture]
.WillOnce(Invoke([kPayloadLength, kFirstSequenceNumber, kFirstTimestamp,
^
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc:317:25: error: lambda capture 'kFirstReceiveTime' is not required to be captured for this use [-Werror,-Wunused-lambda-capture]
kFirstReceiveTime](const SdpAudioFormat& format,
BUG=webrtc:7107
Review-Url: https://codereview.webrtc.org/2672823002
Cr-Commit-Position: refs/heads/master@{#16422}
Left shifting negative integers is undefined behavior, and should be prevented. This CL fixes one such instance in the NetEq Expand function.
BUG=chromium:677106
Review-Url: https://codereview.webrtc.org/2616363003
Cr-Commit-Position: refs/heads/master@{#15966}
The file was aldready pruned down to the point where it only included
webrtc/typedefs.h. Therefore, all includes of
voice_engine_configurations.h are replaced with typedefs.h, except on
two occasions where it was obvously not needed.
BUG=webrtc:6506
Review-Url: https://codereview.webrtc.org/2553583002
Cr-Commit-Position: refs/heads/master@{#15547}
There's no longer any need to make the two arguments have the same
signedness, so we can remove a bunch of superfluous (and sometimes
dangerous) casts.
It turned out I also had to fix the safe_cmp functions to properly handle
enums that are implicitly convertible to integers.
NOPRESUBMIT=true
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2534683002
Cr-Commit-Position: refs/heads/master@{#15281}
There's no longer any need to make the two arguments have the same
signedness, so we can drop the "u" suffix on literal integer
arguments.
NOPRESUBMIT=true
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2535593002
Cr-Commit-Position: refs/heads/master@{#15280}
This is undefined behavior, according to specification.
BUG=chromium:661133
Review-Url: https://codereview.webrtc.org/2500953003
Cr-Commit-Position: refs/heads/master@{#15078}
This can happen in rare and strange cases.
Also taking the opportunity to replace all asserts with DCHECKs in
that file.
BUG=chromium:659225
Review-Url: https://codereview.webrtc.org/2499013002
Cr-Commit-Position: refs/heads/master@{#15070}
With this change, instead of
RTC_DCHECK_GE(unsigned_var, 17u);
we can simply write
RTC_DCHECK_GE(unsigned_var, 17);
or even
RTC_DCHECK_GE(unsigned_var, -17); // Always true.
and the mathematically sensible thing will happen.
Perhaps more importantly, we can replace checks like
// index is size_t, num_channels is int.
RTC_DCHECK(num_channels >= 0
&& index < static_cast<size_t>(num_channels));
or, even worse, just
// Surely num_channels isn't negative. That would be absurd!
RTC_DCHECK_LT(index, static_cast<size_t>(num_channels));
with simply
RTC_DCHECK_LT(index, num_channels);
In short, you no longer have to keep track of the signedness of the arguments, because the sensible thing will happen.
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2459793002
Cr-Commit-Position: refs/heads/master@{#14878}
The timestamps are 32 bit and can (conceivably) be spaced far enough
apart for the calculation, which is done in Q4, to overflow.
BUG=chromium:653268
Review-Url: https://codereview.webrtc.org/2460393002
Cr-Commit-Position: refs/heads/master@{#14856}
PacketList is now list<Packet> instead of list<Packet*>.
Splicing the lists in NetEqImpl::InsertPacketInternal instead of
moving packets. Avoid moving the packet when doing Rfc3389Cng.
Removed PacketBuffer::DeleteFirstPacket and DeleteAllPackets.
BUG=chromium:657300
Review-Url: https://codereview.webrtc.org/2425223002
Cr-Commit-Position: refs/heads/master@{#14747}
Only three items in the (rather large) header were actually used after
InsertPacket: payloadType, timestamp and sequenceNumber. They are now
put directly into Packet. This saves 129 bytes per Packet that no
longer need to be allocated and deallocated.
This also works towards decoupling NetEq from RTP. As part of that,
I've moved the NACK code earlier in InsertPacketInternal, together
with other things that directly reference the RTPHeader.
BUG=webrtc:6549
Review-Url: https://codereview.webrtc.org/2411183003
Cr-Commit-Position: refs/heads/master@{#14658}
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}
This CL deletes the old and not used video defines in
engine_configurations.h and pre-pends voice_ to indicate there are only
voice/audio defines left in the file.
BUG=none
R=solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/2401673002 .
Cr-Commit-Position: refs/heads/master@{#14558}
The original CL (https://codereview.webrtc.org/2315633002) was
reverted since the fuzzer depended on gflags and files in the
resources folder; neither of this is allowed for a fuzzer test in
Chromium. This new version streamlines the dependencies, and changes
the test to generate a sinusoid input audio signal instead of reading
from a file.
Original commit message:
This CL introduces a new fuzzer target neteq_rtp_fuzzer that
manipulates the RTP header fields before inserting the packets into
NetEq. A few helper classes are also introduced.
BUG=webrtc:5447
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device
Review-Url: https://codereview.webrtc.org/2384423002
Cr-Commit-Position: refs/heads/master@{#14523}
NetEq already uses SdpAudioFormat internally; this CL adds an
AudioCodingModule::RegisterReceiveCodec overload that accepts
SdpAudioFormat, and propagates it through AcmReceiver into NetEq.
The intention is to get rid of the other ways to specify decoders and
always use SdpAudioFormat. (And eventually to do the same for encoders
too.)
NOTRY=true
BUG=5801
Review-Url: https://codereview.webrtc.org/2365653004
Cr-Commit-Position: refs/heads/master@{#14506}
Addresses a regression in the NetEq performance test.
# Added NOTRY due to android_arm64_rel being swamped.
NOTRY=True
BUG=chromium:651426
Review-Url: https://codereview.webrtc.org/2383723002
Cr-Commit-Position: refs/heads/master@{#14495}
The RtcEventLog headers need to be accessible from any place which needs
logging, and the implementation needs access to data structures that are
logged.
After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future).
The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/).
This change allows using RtcEventLog in the p2p/ directory, so that we
can log STUN pings and ICE state transitions.
BUG=webrtc:6393
R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2380683005 .
Cr-Commit-Position: refs/heads/master@{#14485}
NetEqDecoder is still used in the external interfaces, but this change
opens up the ability to use SdpAudioFormats directly, once appropriate
interfaces have been added.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2355503002
Cr-Commit-Position: refs/heads/master@{#14368}
The biggest change to NetEq is the move from a primary flag, to a
Priority with two separate levels: one set by RED splitting and one
set by the codec itself. This allows us to unambigously prioritize
"fallback" packets from these two sources. I've chosen what I believe
is the sensible ordering: packets that the codec prioritizes are
chosen first, regardless of if they are secondary RED packets or
not. So if we were to use Opus w/ FEC in RED, we'd only do Opus FEC
decoding if there was no RED packet that could cover the time slot.
With this change, PayloadSplitter now only deals with RED
packets. Maybe it should be renamed RedPayloadSplitter?
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2342443005
Cr-Commit-Position: refs/heads/master@{#14347}
