This changes most non-test related rtc_source_set targets to be
rtc_static_library instead. Targets without any .cc files are excluded.
This should bring back the build behavior we used to have with GYP
(i.e. same symbols exported in the libjingle_peerconnection.a file, which
are used by some downstream projects).
After doing an Android build with these changes:
$ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf
00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory
$ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf
00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*)
00000001 T webrtc::CreatePeerConnectionFactory()
See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries
for more details on this.
NOTICE: This should be further cleaned up in the future, to reduce
binary bloat and unnecessary linking time. Right now it's more
important to restore the desired build output though.
BUG=webrtc:6410, chromium:630755
Review-Url: https://codereview.webrtc.org/2361623004
Cr-Commit-Position: refs/heads/master@{#14364}
This is to fix an issue introduced with iOS 10 where all applications that access the microphone have to include a string in the Info.plist file explaining why they need it.
BUG=webrtc:6403
Review-Url: https://codereview.webrtc.org/2359863003
Cr-Commit-Position: refs/heads/master@{#14354}
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}
1. Remove legacy screen-saver-blocking logic
2. tls_index_ is not a good choice, we can use thread-static
3. ScreenCapturerHelper is not designed for this scenario
4. Disable this capturer on 2+ monitors system
BUG=638802
Review-Url: https://codereview.webrtc.org/2319383002
Cr-Commit-Position: refs/heads/master@{#14342}
1. It looks like ComPtr cannot work well with vector::emplace_back, I got a
consistent crash on one of my machine, but not the other. Move constructor
should have no impact to lvalue reference, but I may be wrong here. The
impact here is ComPtr released before it should be. So a simple solution is to
use copy instead of reference. The D3dDevice is a collection of reference
counted pointers (Microsoft::WRL::ComPtr), there is almost no extra cost.
2. Actively set several fields in D3D11_TEXTURE2D_DESC to avoid potential break
if there are some platform changes later.
3. AcquireNextFrame returns both a DXGI_OUTDUPL_FRAME_INFO with
AccumulatedFrames and an IDXGIResource. But there is no comment in MSDN to
ensure IDXGIResource won't be nullptr if AccumulatedFrames > 0. Adding an extra
check in DxgiOutputDuplicator makes it a safer.
BUG=314516
Review-Url: https://codereview.webrtc.org/2345163002
Cr-Commit-Position: refs/heads/master@{#14341}
On retina display, when we do screen capture, the mouse cursor
looks too small. The reason is that we painted the cursor with
low resolution on to the frame directly.
This CL fixes the bug.
BUG=632995
Review-Url: https://codereview.webrtc.org/2350743003
Cr-Commit-Position: refs/heads/master@{#14340}
This change uses RgbaColor in DesktopFrameGenerator instead of raw uint32_t to
avoid potential endian issues.
BUG=633802
Review-Url: https://codereview.webrtc.org/2334853002
Cr-Commit-Position: refs/heads/master@{#14337}
audio_decoder.cc depends on LegacyEncodedAudioFrame and
LegacyEncodedAudioFrame depends on AudioDecoder::EncodedAudioFrame, so
there's no clear way to separate them as of now. This error is also
hodling up builds downstream. I expect we'll revisit these
dependencies as part of the upcoming larger restructuring effort.
NOTRY=true
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2359763002
Cr-Commit-Position: refs/heads/master@{#14329}
There's still some code run specifically for Opus w/ FEC. It will be
addressed in a separate CL.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2326003002
Cr-Commit-Position: refs/heads/master@{#14319}
Deleted from the VideoFrameBuffer base class.
BUG=webrtc:5921
Review-Url: https://codereview.webrtc.org/2278883002
Cr-Commit-Position: refs/heads/master@{#14317}
- Split out reading/writing of FEC headers to classes separate
from ForwardErrorCorrection. This makes ForwardErrorCorrection
oblivious to what FEC header scheme is used, and lets it focus on
encoding/decoding the FEC payloads.
- Add unit tests for FEC header readers/writers.
- Split ForwardErrorCorrection::XorPackets into XorHeaders and
XorPayloads and reuse these functions for both encoding and
decoding.
- Rename AttemptRecover -> AttemptRecovery in ForwardErrorCorrection.
BUG=webrtc:5654
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2260803002 .
Cr-Commit-Position: refs/heads/master@{#14316}
- Rename GenerateFec -> EncodeFec in ForwardErrorCorrection. This naming
is more consistent with DecodeFec.
- Add appropriate using directives, to reduce clutter in tests.
- Move ConstructMediaPackets to fec_test_helper.{h,cc}. This will help
future tests of ULPFEC/FlexFEC header formatters.
- Generalize tests in rtp_fec_unittest.cc to typed tests. This will help
testing ForwardErrorCorrection with both ULPFEC and FlexFEC.
This CL should not impact functionality or performance.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2267393002
Cr-Commit-Position: refs/heads/master@{#14314}
Before this CL, the IntelligibilityEnhancer introduced a processing delay to the lower band, without compensating for it in the higher bands. This CL corrects this.
BUG=b/30780909
R=henrik.lundin@webrtc.org, peah@webrtc.org
Review URL: https://codereview.webrtc.org/2320833002 .
Cr-Commit-Position: refs/heads/master@{#14311}
Verifies that NetEq doesn't enter muted state when CNG mode is active
and the packet stream is suspended for a long time.
BUG=webrtc:5608
Review-Url: https://codereview.webrtc.org/2335343011
Cr-Commit-Position: refs/heads/master@{#14308}
ssrc taken from packet instead of module removing extra lock
removed unneccesary call to clock_
reduced number of lines.
BUG=webrtc:5565
R=brandtr@webrtc.org
Review URL: https://codereview.webrtc.org/2352023002 .
Cr-Commit-Position: refs/heads/master@{#14307}
This includes if RTCP is received, but the number of packets received by the
other end hasn't increased.
Further, if no RTCP is received for more than 3 feedback intervals (3 seconds)
we start reducing the estimate by 20%. This is put under an experiment.
BUG=webrtc:6238
R=terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2262213002 .
Cr-Commit-Position: refs/heads/master@{#14306}
Trivial patch which fixes an issue where logged rate estimates could be
invalid. E.g. on iOS, two successive timer interrupts can be ~10.5 seconds
and not exactly 10.0 (which is usually the case on Android). In those
cases we could log a rate estimate of e.g. ~51000Hz instead of ~48000Hz.
This CL fixes that.
BUG=NONE
Review-Url: https://codereview.webrtc.org/2350103002
Cr-Commit-Position: refs/heads/master@{#14305}
It requires a new NetEq method, but it can no longer fail. And we no
longer need to use AcmReceiver::decoders_, which we're trying to
eliminate.
(This is a re-land of https://codereview.webrtc.org/2342313002.)
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2348233002
Cr-Commit-Position: refs/heads/master@{#14304}
This change adds a new statistic for logging how many calls to
NetEq::GetAudio resulted in a "muted output". A muted output happens
if the packet stream has been dead for some time (and the last decoded
packet was not comfort noise).
BUG=webrtc:5606
BUG=b/31256483
Review-Url: https://codereview.webrtc.org/2341293002
Cr-Commit-Position: refs/heads/master@{#14302}
It allows the decoder to split the input up into usable chunks before
they are put into NetEq's PacketBuffer. Eventually, all packet splitting
will move into ParsePayload.
There's currently a base implementation of ParsePayload. It will
generate a single Frame that calls the underlying AudioDecoder for
getting Duration() and to Decode.
BUG=webrtc:5805
BUG=chromium:428099
Review-Url: https://codereview.webrtc.org/2326953003
Cr-Commit-Position: refs/heads/master@{#14300}
NOTE: the new code is disabled by default in the WebRtcAudioManager to ensure that
OpenSL ES is not accidentally activated in existing clients. There are still some
unresolved issues to sort out before it can be utilized.
Enables possibility to use OpenSL ES based audio in both directions for WebRTC.
All unit tests and demo clients have been tested with the new implementation but
the new support is behind a flag (see above).
More testing is needed before it can be used in the field and additional support for
hardware effects is still missing.
BUG=webrtc:5925
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2119633004 .
Cr-Commit-Position: refs/heads/master@{#14290}
In the migration to GN templates, some targets got the whole
rtc_common_config removed, which can have unpredicted consequences
in terms of different code behavior due to defines not being set
as expected etc.
It's better to enable this config and only disable the warnings
that fails the build.
BUG=webrtc:6306,webrtc:6307,webrtc:6308
NOTRY=True
Review-Url: https://codereview.webrtc.org/2347263002
Cr-Commit-Position: refs/heads/master@{#14280}
Reason for revert:
Seems to have broken Chromium tests.
Original issue's description:
> AcmReceiver: Look up last decoder in NetEq's table of decoders
>
> AcmReceiver::decoders_ is now one step closer to being unused.
>
> BUG=webrtc:5801
>
> Committed: https://crrev.com/1e4d8b574cde64d93b98d89c7b817fb93185a307
> Cr-Commit-Position: refs/heads/master@{#14274}
TBR=ossu@webrtc.org,henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2348123002
Cr-Commit-Position: refs/heads/master@{#14279}
Reason for revert:
Seems to have broken Chromium tests.
Original issue's description:
> AcmReceiver: Ask NetEq to delete all decoders at once instead of one by one
>
> It requires a new NetEq method, but it can no longer fail. And we no
> longer need to use AcmReceiver::decoders_, which we're trying to
> eliminate.
>
> BUG=webrtc:5801
>
> Committed: https://crrev.com/f6232b43a176e1717354b671a0a52b887d70de59
> Cr-Commit-Position: refs/heads/master@{#14275}
TBR=ossu@webrtc.org,henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2349973002
Cr-Commit-Position: refs/heads/master@{#14278}
Reason for revert:
Seems to have broken Chromium tests.
Original issue's description:
> AcmReceiver::DecoderByPayloadType: Ask NetEq for decoder
>
> Instead of looking in AcmReceiver::decoders_, which we're trying to
> get rid of.
>
> BUG=webrtc:5801
>
> Committed: https://crrev.com/07772e4738ef8007280f97a0245eef34b9ca9391
> Cr-Commit-Position: refs/heads/master@{#14276}
TBR=ossu@webrtc.org,henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2346173002
Cr-Commit-Position: refs/heads/master@{#14277}
