The explicitly defined constructor suppresses the assignment operator,
which blocks the chromium roll.
Bug: b/198565646
Change-Id: I35917d4b99ad86dcf8b9863e798f5a63d9073824
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231123
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34904}
This reverts commit eb89027733c511962120a5f7fd309d1893ad389c.
Reason for revert: We got a successful WebRTC roll into Chromium at last. Relanding, as the issue should be fixed in Chromium by now.
TBR=hta@webrtc.org,philipp.hancke@googlemail.com
Original change's description:
> Revert "frame transformer: make GetPayloadType pure virtual again"
>
> This reverts commit 209ac5fd95594ab3834dad3e3dbd14c8196637bc.
>
> Reason for revert: Breaks WebRTC autoroll presubmit:
> https://chromium-review.googlesource.com/c/chromium/src/+/3134502
> Example failure https://ci.chromium.org/ui/p/chromium/builders/try/mac-rel/775468/overview
>
> ../../buildtools/third_party/libc++/trunk/include/__memory/unique_ptr.h:725:32: error: allocating an object of abstract class type 'testing::NiceMock<blink::(anonymous namespace)::MockTransformableVideoFrame>'
> return unique_ptr<_Tp>(new _Tp(_VSTD::forward<_Args>(__args)...));
> ^
> ../../third_party/blink/renderer/platform/peerconnection/rtc_encoded_video_stream_transformer_test.cc:69:26: note: in instantiation of function template specialization 'std::make_unique<testing::NiceMock<blink::(anonymous namespace)::MockTransformableVideoFrame>>' requested here
> auto mock_frame = std::make_unique<NiceMock<MockTransformableVideoFrame>>();
> ^
> ../../third_party/webrtc/api/frame_transformer_interface.h:36:19: note: unimplemented pure virtual method 'GetPayloadType' in 'NiceMock'
> virtual uint8_t GetPayloadType() const = 0;
> ^
>
>
> Original change's description:
> > frame transformer: make GetPayloadType pure virtual again
> >
> > after chrome was updated in
> > https://chromium-review.googlesource.com/c/chromium/src/+/3103323
> >
> > BUG=webrtc:13077
> >
> > Change-Id: I7e5ff6aaae81c5dcfbaa41b09ef01bc95bb7251a
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230143
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> > Cr-Commit-Position: refs/heads/main@{#34877}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:13077
> Change-Id: I6b2e4e2804890c857f1f832a6a4faa614ec026c4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230920
> Reviewed-by: Olga Sharonova <olka@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Olga Sharonova <olka@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34891}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:13077
Change-Id: I8414f74be87aad62166a95fac0cd400257fd25a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231120
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34901}
This reverts commit fb0dca6c055cbf9e43af665d3c437eba6f43372e.
Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium.
Original change's description:
> Wire up non-sender RTT for audio, and implement related standardized stats.
>
> The implemented stats are:
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
>
> Bug: webrtc:12951, webrtc:12714
> Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34861}
# Not skipping CQ checks because original CL landed > 1 day ago.
TBR=hta,hbos,minyue
Bug: webrtc:12951, webrtc:12714
Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34897}
VideoDecoder no longer uses this VideoCodec class,
thus this member is unused.
Bug: webrtc:13045
Change-Id: I6e46a563e90f2538bf288995a3837d95c00ba9cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230941
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34896}
This reverts commit 209ac5fd95594ab3834dad3e3dbd14c8196637bc.
Reason for revert: Breaks WebRTC autoroll presubmit:
https://chromium-review.googlesource.com/c/chromium/src/+/3134502
Example failure https://ci.chromium.org/ui/p/chromium/builders/try/mac-rel/775468/overview
../../buildtools/third_party/libc++/trunk/include/__memory/unique_ptr.h:725:32: error: allocating an object of abstract class type 'testing::NiceMock<blink::(anonymous namespace)::MockTransformableVideoFrame>'
return unique_ptr<_Tp>(new _Tp(_VSTD::forward<_Args>(__args)...));
^
../../third_party/blink/renderer/platform/peerconnection/rtc_encoded_video_stream_transformer_test.cc:69:26: note: in instantiation of function template specialization 'std::make_unique<testing::NiceMock<blink::(anonymous namespace)::MockTransformableVideoFrame>>' requested here
auto mock_frame = std::make_unique<NiceMock<MockTransformableVideoFrame>>();
^
../../third_party/webrtc/api/frame_transformer_interface.h:36:19: note: unimplemented pure virtual method 'GetPayloadType' in 'NiceMock'
virtual uint8_t GetPayloadType() const = 0;
^
Original change's description:
> frame transformer: make GetPayloadType pure virtual again
>
> after chrome was updated in
> https://chromium-review.googlesource.com/c/chromium/src/+/3103323
>
> BUG=webrtc:13077
>
> Change-Id: I7e5ff6aaae81c5dcfbaa41b09ef01bc95bb7251a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230143
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/main@{#34877}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:13077
Change-Id: I6b2e4e2804890c857f1f832a6a4faa614ec026c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230920
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34891}
This patch adds support for manually setting subnets that
should be handled as VPN, i.e be subject to VpnPreference,
in case webrtc fails to auto-detect VPNs.
Bug: webrtc:13097
Change-Id: I42514f0677a35cfe30ad053570fa9c2a5b4a856b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230122
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34852}
This patch adds a vp preference field to RTCConfig.
DEFAULT, // No VPN preference.
ONLY_USE_VPN, // only use VPN connections.
NEVER_USE_VPN, // never use VPN connections
PREFER_VPN, // use a VPN connection if possible, i.e VPN connections sorts higher than all other connections.
AVOID_VPN, // only use VPN if there is no other connections, i.e VPN connections sorts last.
Bug: webrtc:13097
Change-Id: I3f95bdfa9134e082c7d389f803bd08facfb70262
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229591
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34842}
All decoders are supposed to be able to decode all valid bitstreams
that can be produced by an encoder. In the cases where this is not
the case, reference_scaling better captures the cause of this than
scalability_mode which was used initially.
Bug: chromium:1187565
Change-Id: I21174077badf0fb9d90b1b58f003edac5b8ee0f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229184
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34800}
Setting different number of temporal layers is supported by SimulcastEncodeAdapter and LibvpxVp8Encoder will fallback to SimulcastEncoderAdapter if InitEncode fails.
Bug: none
Change-Id: I8a09ee1e6c70a0006317957c0802d019a0d28ca2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228642
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34785}
from runtime check in proxy classes that picks decoder (VCMDecoderDataBase)
to a DCHECK in the VideoDecoder::Settings
Bug: None
Change-Id: Ic8c2e971486a3a7eb247f9d03815aba5ca5a7bad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228644
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34761}
GICE was removed around M42
BUG=webrtc:4299
Change-Id: I4e83a888c3ecc1681799c07b47b75c9f31b40baa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227348
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34680}
Some hardware H.264 encoders does not place average QP delta in
slice_qp_delta field. Adding an optional flag in EncoderInfo to notify
quality scaler about this.
Bug: webrtc:12942
Change-Id: I3ee29c5ae9bd7bb34d26eba7e6bede3798ca44b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226921
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34627}
Uppercase constants are more likely to conflict with macros (for
example rtc::SRTP_AES128_CM_SHA1_80 and OpenSSL SRTP_AES128_CM_SHA1_80).
This CL renames some constants and follows the C++ style guide.
Bug: webrtc:12997
Change-Id: I2398232568b352f88afed571a9b698040bb81c30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226564
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34553}
Just like the C++ API, add a method in Java VideoFrame.Buffer that
describes the underlying implementation.
Use this method to properly select AndroidVideoBuffer
or AndroidVideoI420Buffer in Java -> C++ Video Frame Conversion.
Also, add a test case for WrappedNativeI420Buffer
in VideoFrameBufferTest for consistency.
Bug: webrtc:12602
Change-Id: I4c0444e8af6f6a1109bc514e7ab6c2214f1f6d60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223080
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34545}
As part of go/coil update code search links to not point to the
"master" branch.
Bug: chromium:1226942
Change-Id: I0ae9e84ecc660f789a69fe0b226f93bbc39a8a66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226081
Commit-Queue: Tony Herre <toprice@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34531}
This CL completes the removal of assert() and relative headers from
the codebase (excluded
//examples/objc/AppRTCMobile/third_party/SocketRocket which is in a
third_party sub-directory).
Bug: webrtc:6779
Change-Id: I93ed57168d2c0e011626873d66529488c5f484f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225546
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34528}
Software fallback wrapper now reports least common multiple of requirements
for two encoders.
SimulcastEncoderAdapter queries actual encoder before InitEncode call
and requests alignment for all layers if simulcast is not supported by
any of the encoders.
Bug: chromium:1084702
Change-Id: Iaed8190737125d447036b6c664b863be72556a5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225881
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34466}
CL partially auto-generated with:
git grep -l "\bassert(" | grep "\.[c|h]" | \
xargs sed -i 's/\bassert(/RTC_DCHECK(/g'
And with:
git grep -l "RTC_DCHECK(false)" | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/g'
With some manual changes to include "rtc_base/checks.h" where
needed.
A follow-up CL will remove assert() from Obj-C code as well
and remove the #include of <assert.h>.
The choice to replace with RTC_DCHECK is because assert()
is because RTC_DCHECK has similar behavior as assert()
based on NDEBUG.
This CL also contains manual changes to switch from
basic RTC_DCHECK to other (preferred) versions like
RTC_DCHECK_GT (and similar).
Bug: webrtc:6779
Change-Id: I00bed8886e03d685a2f42324e34aef2c9b7a63b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224846
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34442}
This is needed to facilitate dumping of stats to CSV in tests.
Bug: none
Change-Id: Ic78a4630f70a9238d26161ac89c205903dfc852f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225300
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34435}
