This reduces binary size considerably and solves some other problems.
Also rewrote using variadic templates.
Initial patch contributed by andrey.semashev@gmail.com.
BUG=webrtc:2305
Review-Url: https://codereview.webrtc.org/2509733003
Cr-Commit-Position: refs/heads/master@{#16703}
This change adds a flag, use_sctpmap, to DataContentDescription. The deserialization code sets the flag based on the format of the m= line.
There were already unit tests using SDP in the new format, so I just updated them to check use_sctpmap was set as expected.
The change to mediasession copies use_sctpmap from the offered DataContentDescription to the answer.
I haven't figured out how to test this change yet, but wanted to get feedback before continuing.
BUG=chromium:686212
Review-Url: https://codereview.webrtc.org/2690943011
Cr-Commit-Position: refs/heads/master@{#16686}
The AsyncClosures only ever have one thing referencing them, so they
should be using std::unique_ptr to manage ownership. Maybe this code was
written before std::unique_ptr was available.
Originally reverted because it made a change to ScopedMessageData
that wasn't backwards compatible, and applications using the rtc::Thread
infrastructure may be using it.
BUG=None
NOTRY=True
Review-Url: https://codereview.webrtc.org/2689233003
Cr-Commit-Position: refs/heads/master@{#16684}
Reason for revert:
The change to messagequeue.h isn't backwards compatible. Will reland after making it backwards compatible.
Original issue's description:
> Use std::unique_ptr instead of rtc::scoped_refptr in AsyncInvoker.
>
> The AsyncClosures only ever have one thing referencing them, so they
> should be using std::unique_ptr to manage ownership. Maybe this code was
> written before std::unique_ptr was available.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2689233003
> Cr-Commit-Position: refs/heads/master@{#16680}
> Committed: a5a472927bTBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2703613006
Cr-Commit-Position: refs/heads/master@{#16683}
The first unsignalled SSRC creates a default receive channel.
Any unsignalled SSRC changes after that replace the default SSRC.
Add unit tests for changing unsignalled SSRCs.
BUG=webrtc:5208
Review-Url: https://codereview.webrtc.org/2692993009
Cr-Commit-Position: refs/heads/master@{#16682}
The AsyncClosures only ever have one thing referencing them, so they
should be using std::unique_ptr to manage ownership. Maybe this code was
written before std::unique_ptr was available.
BUG=None
Review-Url: https://codereview.webrtc.org/2689233003
Cr-Commit-Position: refs/heads/master@{#16680}
This implementation uses various legacy classes such as EventTimeWrapper,
CriticalSectionWrapper, EventWrapper etc and hasn't been maintained
(or used?) for a long time.
Instead of spending time on testing and updating the class, I think
we should just remove it. For versions of Windows that we support,
following Win7, we use the CoreAudio implementation.
BUG=webrtc:7183
R=solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2700983002 .
Cr-Commit-Position: refs/heads/master@{#16678}
FallbackDesktopCapturerWrapper is a DesktopCapturer implementation, which owns
two DesktopCapturer implementations. If the main DesktopCapturer fails, it uses
the secondary capturer. The logic is now used in ScreenCapturerWinMagnifier, and
it can also be shared in ScreenCapturerWinDirectx to fallback to Gdi capturer on
privilege prompt or login screen.
BUG=684937
Review-Url: https://codereview.webrtc.org/2697453002
Cr-Commit-Position: refs/heads/master@{#16677}
This does not fix the myriad of other problems here, but at least
removes the dependency on CONF_VALUE.
BUG=526270
Review-Url: https://codereview.webrtc.org/2705603003
Cr-Commit-Position: refs/heads/master@{#16676}
The url of the ICE server is added to the IceCandiate class.
This can be used to tell which server this candidate was gathered from.
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2690593002
Cr-Commit-Position: refs/heads/master@{#16675}
This is necessary in case the drawer doesn't cover all the pixels.
BUG=None
Review-Url: https://codereview.webrtc.org/2704663002
Cr-Commit-Position: refs/heads/master@{#16671}
and the method RTPSender::GenerateNewSSRC.
It's now mandatory for higher layers to call SetSSRC, RTPSender
no longer allocates any ssrc by default.
BUG=webrtc:4306,webrtc:6887
Review-Url: https://codereview.webrtc.org/2644303002
Cr-Commit-Position: refs/heads/master@{#16670}
This a pre-step for improving perfomance of the RTCPReceiver
- rest of the ReceiveInfo is tmmbr related and
can be handled only when tmmbr is explicitly enabled.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2681003003
Cr-Commit-Position: refs/heads/master@{#16667}
This feature is unused. We can then also delete the header file
video_capture_delay.h.
BUG=None
Review-Url: https://codereview.webrtc.org/2665113006
Cr-Commit-Position: refs/heads/master@{#16666}
In order to not make this CL too large I have broken it down into at least two steps. In this CL we only propagate the pacing information part of the way:
webrtc::PacedSender::Process <--- propagate from here
webrtc::PacedSender::SendPacket
webrtc::PacketRouter::TimeToSendPacket
webrtc::ModuleRtpRtcpImpl::TimeToSendPacket <--- to here
webrtc::RTPSender::TimeToSendPacket
webrtc::RTPSender::PrepareAndSendPacket
webrtc::RTPSender::AddPacketToTransportFeedback
webrtc::TransportFeedbackAdapter::AddPacket
webrtc::SendTimeHistory::AddAndRemoveOld <--- goal is to propagte it here
BUG=webrtc:6822
Review-Url: https://codereview.webrtc.org/2628563003
Cr-Commit-Position: refs/heads/master@{#16664}
Multimedia timers are higher precision than WM_TIMER, but they're also
a limited resource and more costly. So this implementation is a best
effort implementation that falls back on WM_TIMER when multimedia
timers aren't available.
A possible future change could be to make high precision timers in a
TaskQueue, optional. The reason for doing so would be for TaskQueues
that don't need high precision timers, won't eat up timers from TQ
instances that really need it.
BUG=webrtc:7151
Review-Url: https://codereview.webrtc.org/2691973002
Cr-Commit-Position: refs/heads/master@{#16661}
This utility class can be used to represent either an error or a
successful return value. Follows the pattern of StatusOr in the protobuf
library.
This will be used by ORTC factory methods; for instance, CreateRtpSender
will either return an RtpSender or an error if the parameters are
invalid or some other failure occurs.
This CL also moves RTCError classes to a separate file, and adds tests
that were missing before.
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/2692723002
Cr-Commit-Position: refs/heads/master@{#16659}
This change adds a ResolutionChangeDetector to help dxgi components, say
DxgiDuplicatorController and DxgiTexture to detect resolution changes.
BUG=684162
Review-Url: https://codereview.webrtc.org/2682913002
Cr-Commit-Position: refs/heads/master@{#16654}
The url of the ICE server is added to the IceCandiate class.
This can be used to tell which server this candidate was gathered from.
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2688943003
Cr-Commit-Position: refs/heads/master@{#16652}
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}
The codecs expected by HasCorrectCodecs now depends which codecs were
enabled by build flags.
SendSideBweWithOverheadTest.MinAndMaxBitrate now expects different
values for min bitrate depending on if we support 120 ms frames for
Opus.
BUG=b/35415435
Review-Url: https://codereview.webrtc.org/2691343008
Cr-Commit-Position: refs/heads/master@{#16643}
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}
The frame_analyzer which is used by compare_videos.py needs to handle
barcode errors. As before the reference and the test video can contain
repeated frames. When there are barcode decode errors in the test video,
then we should not let that contribute to the skipped frames score. When
there are barcode decode errors in the reference video, then we need to
take proper care to still calculate skipped barcodes when the
corresponding frames are not present in the test video and the test
video does not have a frame with a barcode decode error that could have
been the same frame as the one in the reference.
A new metric total number of skipped frames and for number of decode
errors is introduced. Barcodes that appears in the test video, but not
in the reference video are also listed.
BUG=webrtc:6967
Review-Url: https://codereview.webrtc.org/2666333003
Cr-Commit-Position: refs/heads/master@{#16638}
Previously, was only checking the Android SDK version. But it also needs
to check for the presence of the connectivity manager service.
BUG=webrtc:7026
Review-Url: https://codereview.webrtc.org/2697943002
Cr-Commit-Position: refs/heads/master@{#16631}
No point in measuring the time needed to write dropped frames to disk.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2696503003
Cr-Commit-Position: refs/heads/master@{#16629}
For them implemeted upscaling in libyuv metrics calculation.
Updated maximum number of SL in vp9 encoder to 3.
Refactored names of some fields in Video_quality_check analyzer.
BUG=webrtc:7095
Review-Url: https://codereview.webrtc.org/2681683003
Cr-Commit-Position: refs/heads/master@{#16625}
Add explicit conversions to/from uint64_t
uint64_t is natural type for NtpTime, including wrap on overflow.
BUG=None
Review-Url: https://codereview.webrtc.org/2695683002
Cr-Commit-Position: refs/heads/master@{#16624}
The removed tests are covered by cases in call_perf_tests.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2672583002
Cr-Commit-Position: refs/heads/master@{#16621}
