In addition to setting or appending from another Buffer, which was
already possible, this allows for e.g. std::vector and rtc::ArrayView
arguments.
Review-Url: https://codereview.webrtc.org/2293983002
Cr-Commit-Position: refs/heads/master@{#14073}
this eliminates reparsing of rtp packet on send audio path
BUG=webrtc:5261
Review-Url: https://codereview.webrtc.org/2292883002
Cr-Commit-Position: refs/heads/master@{#14072}
Reason for revert:
ScreenCapturerTest.CaptureUpdatedRegion fails on Win DrMemory Full.
Original issue's description:
> [WebRTC] A real ScreenCapturer test
>
> We do not have a real ScreenCapturer test before. And after CL 2210443002, a new
> ScreenDrawer interface is added to the code base to draw various shapes on the
> screen. This change is to use ScreenDrawer to test ScreenCapturer. Besides test
> cases, some other changes are included,
>
> 1. A WaitForPendingPaintings() function in ScreenDrawer, to wait for a
> ScreenDrawer to finish all the pending draws. This function now only sleeps 50
> milliseconds on X11 and 100 milliseconds on Windows.
>
> 2. A Color structure to help handle a big-endian or little-endian safe color and
> provide functions to compare with DesktopFrame::data(). Both ScreenDrawer and
> DesktopFrameGenerator (in change 2202443002) can use this class to create colors
> and compare with or paint to a DesktopFrame.
>
> 3. ScreenDrawer now uses Color structure instead of uint32_t.
>
> BUG=314516
>
> TBR=kjellander@chromium.org
>
> Committed: https://crrev.com/9d1c54ace0dc9f68da0152aa1ded2a8dba0a43ae
> Cr-Commit-Position: refs/heads/master@{#14058}
TBR=sergeyu@chromium.org,jamiewalch@chromium.org,kjellander@chromium.org,zijiehe@chromium.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=314516
Review-Url: https://codereview.webrtc.org/2310953002
Cr-Commit-Position: refs/heads/master@{#14071}
We need this for the bots to be functional in client.webrtc.fyi.
Right now they're not really working since runhooks no longer runs GYP
(nor GN), and the MB step is not yet enabled in this waterfall.
BUG=webrtc:6287
NOTRY=True
Review-Url: https://codereview.webrtc.org/2312733002
Cr-Commit-Position: refs/heads/master@{#14070}
Remove common_inherited_config from the targets and add it to the
template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2311843002
Cr-Commit-Position: refs/heads/master@{#14069}
Before this change, the argument still needed those methods, but not
having them resulted in a compilation error. Now, it results in this
constructor being removed from the overload set.
This currently makes no difference, but I'm about to publish a CL that
breaks without this.
Review-Url: https://codereview.webrtc.org/2312473002
Cr-Commit-Position: refs/heads/master@{#14068}
Add "//build/config/compiler:optimize_max" to rtc_add_configs and
"//build/config/compiler:default_optimization" to rtc_remove_configs.
This is the default optimization in GYP, and might help explain a 82.5%
regression in webrtc_perf_tests at 13946:13946
BUG=chromium:641966
NOTRY=True
Review-Url: https://codereview.webrtc.org/2307283002
Cr-Commit-Position: refs/heads/master@{#14067}
Changes GetStatsReport to a callback-based function. Stats collection
is dispatched to three different stats collecting methods, being
invoked asynchronously on the signaling, worker and network threads.
The three resulting stats reports are merged into one before returned.
The only current stats being collected is on the signaling thread, but
a FakeRTCStatsCollector is able to test the multi-threaded and
stats-merging behaviors. Future CLs simply have to put their stats
collecting code in the appropriate ProducePartialResultsOnFooThread
method.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2270033004
Cr-Commit-Position: refs/heads/master@{#14064}
Remove common_config from the targets' config and add
it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2300413002
Cr-Commit-Position: refs/heads/master@{#14063}
Replaces render_time_ms_, but old accessors are kept for
compatibility.
Also short-circuit timestamp translation in
WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame.
BUG=webrtc:5682, webrtc:5740
Review-Url: https://codereview.webrtc.org/2282713002
Cr-Commit-Position: refs/heads/master@{#14062}
We do not have a real ScreenCapturer test before. And after CL 2210443002, a new
ScreenDrawer interface is added to the code base to draw various shapes on the
screen. This change is to use ScreenDrawer to test ScreenCapturer. Besides test
cases, some other changes are included,
1. A WaitForPendingPaintings() function in ScreenDrawer, to wait for a
ScreenDrawer to finish all the pending draws. This function now only sleeps 50
milliseconds on X11 and 100 milliseconds on Windows.
2. A Color structure to help handle a big-endian or little-endian safe color and
provide functions to compare with DesktopFrame::data(). Both ScreenDrawer and
DesktopFrameGenerator (in change 2202443002) can use this class to create colors
and compare with or paint to a DesktopFrame.
3. ScreenDrawer now uses Color structure instead of uint32_t.
BUG=314516
TBR=kjellander@chromium.org
Review-Url: https://codereview.webrtc.org/2268093002
Cr-Commit-Position: refs/heads/master@{#14058}
This test failed on the memcheck bot:
https://build.chromium.org/p/client.webrtc/builders/Linux%20Memcheck/builds/6704/steps/video_engine_tests/logs/stdio
The test assumed that the absolute send time header extension can never
be zero. It's a timestamp truncated to 24 bits, and zero is not a
special value - so it can very rarely end up being precisely zero.
The fix makes the test wait for at least one packet having a non-zero send time.
I've considered changing the test to use a fake clock instead to ensure
that not only the value is non-zero, but that it indeed reflects the
system timestamp - but that involves changing a very large number of
files. Besides, other tests in this file don't verify values for header
extensions where zeroes are allowed.
NOTRY=true
Review-Url: https://codereview.webrtc.org/2307693002
Cr-Commit-Position: refs/heads/master@{#14056}
ParsedRtcEventLog::ParseStream was using a stack-allocated 64kb array
for a temporary buffer. This was causing problems in build environments
with restrictions on stack size.
This change replaces it with an std::vector.
NOTRY=true
Review-Url: https://codereview.webrtc.org/2297343003
Cr-Commit-Position: refs/heads/master@{#14055}
The helpers intended to replace and deprecate BuildRtpHeader when
RtpSenderAudio/RtpSenderVideo will be updated to pass RtpPacket class
instead of raw buffer for sending.
BUG=webrtc:5261
R=sprang@webrtc.org
Review URL: https://codereview.webrtc.org/2303283002 .
Cr-Commit-Position: refs/heads/master@{#14051}
Remove //build/config/sanitizers:deps as a dependency for
all rtc_executable targets and add it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2308553002
Cr-Commit-Position: refs/heads/master@{#14048}
Defines the rtc_executable, rtc_source_set, rtc_test and
rtc_static_library templates.
These templates provide no functionality yet, but will enable common
configuration to be introduced, avoiding repetition in every target
Changes summary:
- Prepend rtc_ to test, source_set, executable and static_library targets
- Change "configs -= [" to "suppressed_configs += ["
- Include webrtc/build/webrtc.gni where it wasn't included yet
- Delete import("//testing/test.gni"), since rtc_test makes it unnecessary.
BUG=webrtc:6187
TBR=henrik.lundin@webrtc.org,tommi@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2301053002
Cr-Commit-Position: refs/heads/master@{#14043}
Reason for revert:
Reland this now that downstream tests have been fixed.
Original issue's description:
> Revert of Add pps id and sps id parsing to the h.264 depacketizer. (patchset #5 id:80001 of https://codereview.webrtc.org/2238253002/ )
>
> Reason for revert:
> Breaks some h264 bitstream tests downstream. Reverting for now.
>
> Original issue's description:
> > Add pps id and sps id parsing to the h.264 depacketizer.
> >
> > BUG=webrtc:6208
> >
> > Committed: https://crrev.com/abcc3de169d8896ad60e920e5677600fb3d40180
> > Cr-Commit-Position: refs/heads/master@{#13838}
>
> TBR=sprang@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6208
>
> Committed: https://crrev.com/83d79cd4a2bfbdd1abc1f75480488df4446f5fe0
> Cr-Commit-Position: refs/heads/master@{#13844}
TBR=sprang@webrtc.org,kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6208
Review-Url: https://codereview.webrtc.org/2302893002
Cr-Commit-Position: refs/heads/master@{#14042}
- Remove webrtc/tools/agc/test_utils.cc/.h - only used from the above test.
- Remove webrtc/tools/agc/agc_harness.cc - not used anymore.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2299023004
Cr-Commit-Position: refs/heads/master@{#14039}
These methods are not used by the new AndroidVideoTrackSource API.
Review-Url: https://codereview.webrtc.org/2280873002
Cr-Commit-Position: refs/heads/master@{#14036}
Reason for revert:
Breaks downstream build.
Original issue's description:
> Ignore Camera and Flip bits in CVO when parsing video rotation
>
> Currently, if WebRTC receives a CVO byte where the Camera or Flip bit is
> set, then rotation is incorrectly parsed as 0. This CL fixes that issue.
> The Camera and Flip bit is still unimplemented and will just be ignored
> though.
>
> BUG=webrtc:6120
> R=danilchap@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
>
> Committed: f9e1b922efTBR=pthatcher@webrtc.org,danilchap@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6120
Review-Url: https://codereview.webrtc.org/2300323002
Cr-Commit-Position: refs/heads/master@{#14035}
The sample uses are from when I debugged bug 617124. The change in neteq_network_stats_unittest.cc is a fix for a minor unrelated bug found by the try bots when I tried to land this CL (a test was passing uninitialized packet data to NetEq).
BUG=chromium:617124
Review-Url: https://codereview.webrtc.org/2293893002
Cr-Commit-Position: refs/heads/master@{#14034}
If a CNG packet is received first, followed by a speech packet with
another sample rate, NetEq should treat this as a change of codec, flush
out the CNG packet and reset the sample rate to that of the speech
packet.
BUG=webrtc:5447
NOTRY=True
Review-Url: https://codereview.webrtc.org/2307493002
Cr-Commit-Position: refs/heads/master@{#14032}
run it is important that the same build flags are used in the code being
tested. For the debugging functionality inside APM, that was not the case
and this is corrected in this CL.
This CL is chained to the CL https://codereview.webrtc.org/2300813004/
BUG=webrtc:5298
Review-Url: https://codereview.webrtc.org/2307563002
Cr-Commit-Position: refs/heads/master@{#14031}
Currently, if WebRTC receives a CVO byte where the Camera or Flip bit is
set, then rotation is incorrectly parsed as 0. This CL fixes that issue.
The Camera and Flip bit is still unimplemented and will just be ignored
though.
BUG=webrtc:6120
R=danilchap@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2280703002 .
Cr-Commit-Position: refs/heads/master@{#14027}
Currently, the aec_debug_dump buildflag can and is used to store data in the whole of
the audio processing module. Therefore a more appropriate name is apm_debug_dump which
also matches the names of the data dumping functionality. This CL makes that name change.
The CL also changes the WEBRTC_AEC_DEBUG_DUMP define to
WEBRTC_APM_DEBUG_DUMP == 1
Furthermore, this CL moves the buildflag to a more appropriate place.
BUG=webrtc:5298
Review-Url: https://codereview.webrtc.org/2300813004
Cr-Commit-Position: refs/heads/master@{#14026}
WebRTC no longer has any restriction on what thread frames should be
delivered on. One possible problem with this CL is that NV21->I420
conversion and scaling is done on the thread that delivers frames, which
might cause fps regressions.
R=nisse@webrtc.org, perkj@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/2137503003 .
Cr-Commit-Position: refs/heads/master@{#14021}
