The webrtc::VideoStreamDecoderInterface was basically created as a public version of FrameBuffer2, but to hide the complexity of FrameBuffer2 it was also combined with decoding so that the public API could be reasonably simple to use. FrameBuffer3 has a simple API with a clear purpose, so its API can be exposed directly.
Bug: webrtc:14026
Change-Id: I81dc84b869e4d16c5e02feb5c876fbcede3d4a25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261181
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36781}
The features have two safety fallbacks:
- multichannel config has a killswitch WebRTC-Aec3SetupSpecificDefaultConfigDefaultsKillSwitch
- stereo detection has a killswitch WebRTC-Aec3StereoContentDetectionKillSwitch
Both features are enabled by default in the AEC3 config.
Tested: Bitexact on a large number of aecdumps.
Bug: chromium:1295710
Change-Id: I340cdc9140dacd4ca22d0911eb9f732b6cf8b226
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258129
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36482}
This explores the theory that targets that have no files, just
dependencies, are unnecessary.
Bug: webrtc:13805
Change-Id: I1feb50cf3886128031af8970eae361e35fb052c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256974
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36363}
This patch takes a stab at modules/video_coding,
but reaches only about half.
Bug: webrtc:10335
Change-Id: I0d47d0468b818145470c51ae4e8e75ff58d499ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256112
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36335}
The problem was fixed by implementing the methid PacketDuration() in
AudioDecoderG722StereoImpl, which catches the issue in
AudioDecoder::Decode().
Bug: chromium:1280851
Change-Id: I31f974b9999f3c1c62b0e5dc39bb3e56a9a9388d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251842
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36034}
This emulates behaviour from frame buffer 2, but does not handle stats.
In contrast to frame buffer 2, all work happens on the same task queue.
FrameBuffer3Proxy encapsulates FrameBuffer3 and scheduler behind
a field trial WebRTC-FrameBuffer3.
This separates frame scheduling behaviour into a few components,
VideoReceiveStreamTimeoutTracker
* Handles the stream timeouts.
FrameDecodeScheduler
* Manages the scheduling and cancelling of frames being sent to the
decoder.
FrameDecodeTiming
* Handles the timing and ordering of frames to be decoded.
Other changes
* Adds CurrentSize() method to FrameBuffer3
* Move timing to a separate library
* Does a thread check for Receive statistics as this is now
on the worker thread.
* Adds `FlushImmediate` method to RunLoop so that
video_receive_stream2_unittest can pass when scheduling is happening
on the worker thread.
Change-Id: Ia8d2e5650d1708cdc1be3631a5214134583a0721
Bug: webrtc:13343
Tested: Ran webrtc_perf_tests, video_engine_tests, rtc_unittests forcing frame buffer3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241603
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35847}
This makes fuzzer test cases fail if there's an assert failure in
the helper functions called by the test.
Bug: None
Change-Id: Ic187d72b8d4e016659a68a7bdcaadb78ab2aab05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246400
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35804}
Important: This change does not in any way affect echo cancellation or standardized stats. The user audio experience is unchanged. Only non-standard stats are affected. Echo return loss metrics are unchanged. Residual echo likelihood {recent max} will no longer be computed by default.
Important: The echo detector is no longer enabled by default.
API change, PSA: https://groups.google.com/g/discuss-webrtc/c/mJV5cDysBDI/m/7PTPBjVHCgAJ
This CL removes the default usage of the residual echo detector in APM.
It can now only be used via injection and the helper function webrtc::CreateEchoDetector. See how the function audio_processing_unittest.cc:CreateApm() changed, for an example.
The echo detector implementation is marked poisonous, to avoid accidental dependencies.
Some cleanup is done:
- EchoDetector::PackRenderAudioBuffer is declared in one target but is defined in another target. It is not necessary to keep in the API. It is made an implementation detail, and the echo detector input is documented in the API.
- The internal state of APM is large and difficult to track. Submodule pointers that are set permanently on construction are now appropriately marked const.
Tested:
- existing + new unit tests
- audioproc_f is bitexact on a large number of aecdumps
Bug: webrtc:11539
Change-Id: I00cc2ee112fedb06451a533409311605220064d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239652
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35550}
FrameBuffer3 keep track of order, decodability and continuity of the inserted frames. Compared to FrameBuffer2 which schedule frames for decoding and is thread safe, FrameBuffer3 does not schedule decoding and is thread unsafe.
Change-Id: Ic3bd540c4f69cec26fce53a40425f3bcd9afe085
Bug: webrtc:13343
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238985
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35494}
This is tested by a simple unit test and a new fuzzer that verify that all that can be parsed also can be written.
Bug: webrtc:12000
Change-Id: I461aedf97d3dec6e8916e72110fa097c3b31c27f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231642
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34986}
The VP9 encoder may drop a frame internally which will not advance the
frame pattern. Consider the following scenario where only spatial layer
0 and temporal layer 0 is active:
1. Key frame encoded
2. Spatial layer 1 is activated
3. Delta T0 dropped
4. Delta T0 encoded
No S1T0 frame is encoded in (1) since it's not active. When
NextFrameConfig is called in (3) it will say that future frames may
reference T0 on both S0 and S1, but it's then dropped.
On step (4), the SVC controller essentially thinks it's encoding a new
picture and will happily reference the T0 on what it thinks is the first
delta frame. However, this is actually still the key frame and since
there was no S1T0 frame produced it will reference an invalid buffer.
To fix this, only say it's possible to reference a T0 frame after it has
been successfully encoded.
Bug: webrtc:11999, webrtc:13142, chromium:1178444
Change-Id: Iab3d2042ce0b3fa7d952b2831d1a36b1a6613a86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231695
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34982}
This is a reland of 9e09831767995531ae1c2804e1c15fa2be4053f2
The field "additional_configs" needs to be used to set "configs"
for the "fuzzer_test" GN template. See
https://source.chromium.org/chromium/chromium/src/+/main:testing/libfuzzer/fuzzer_test.gni;l=18;drc=825f86aa594207bfc50f87495544b48014814c9d.
Original change's description:
> Make webrtc_fuzzer_test use //:common_config.
>
> Before this CL, the GN template webrtc_fuzzer_test was using a build
> config that was different from the one used by other WebRTC's targets.
>
> We discovered this in [1] where we detected that RTC_DCHECK_IS_ON had
> different values across translation units (1 everywhere and 0 in the
> one of the .cc file owned by the webrtc_fuzzer_test).
>
> This was because webrtc_fuzzer_test was not including the default
> config //:common_config in its "configs".
>
> [1] - https://webrtc-review.googlesource.com/c/src/+/226465
>
> Bug: None
> Change-Id: I5635d90281769c23c5d86ebc8cb494da029c2e85
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226540
> Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34509}
Bug: None
Change-Id: I56e2a7ea811a94762e09953acf3d33d3f46b1d24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226542
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34511}
This reverts commit 9e09831767995531ae1c2804e1c15fa2be4053f2.
Reason for revert: The "fuzzer_test" GN template expanded by
"webrtc_fuzzer_test" still ignores the "configs" and another
field needs to be used.
Original change's description:
> Make webrtc_fuzzer_test use //:common_config.
>
> Before this CL, the GN template webrtc_fuzzer_test was using a build
> config that was different from the one used by other WebRTC's targets.
>
> We discovered this in [1] where we detected that RTC_DCHECK_IS_ON had
> different values across translation units (1 everywhere and 0 in the
> one of the .cc file owned by the webrtc_fuzzer_test).
>
> This was because webrtc_fuzzer_test was not including the default
> config //:common_config in its "configs".
>
> [1] - https://webrtc-review.googlesource.com/c/src/+/226465
>
> Bug: None
> Change-Id: I5635d90281769c23c5d86ebc8cb494da029c2e85
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226540
> Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34509}
TBR=mbonadei@webrtc.org,landrey@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: Iec13b411e7f027e78e731e3242e0557b6de38a2b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226541
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34510}
Before this CL, the GN template webrtc_fuzzer_test was using a build
config that was different from the one used by other WebRTC's targets.
We discovered this in [1] where we detected that RTC_DCHECK_IS_ON had
different values across translation units (1 everywhere and 0 in the
one of the .cc file owned by the webrtc_fuzzer_test).
This was because webrtc_fuzzer_test was not including the default
config //:common_config in its "configs".
[1] - https://webrtc-review.googlesource.com/c/src/+/226465
Bug: None
Change-Id: I5635d90281769c23c5d86ebc8cb494da029c2e85
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226540
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34509}
This removes PacketRouter inheritance from RemoteBitrateObserver and TransportFeedbackSenderInterface.
Call binds methods for sending REMB and transport feedback messages from RemoteCongestionController to PacketRouter.
This is needed until the RTCPTranseiver is used instead of the RTP modules.
Bug: webrtc:12693
Change-Id: I7088de497cd6d1e15c98788ff3e6b0a2c8897ea8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215965
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33993}
Fix simulcast svc controller to reuse dropped frame configuration,
same as full svc and k-svc controllers do.
This fuzzer reminded the issue was still there.
This is a reland of https://webrtc-review.googlesource.com/c/src/+/212281
Bug: webrtc:11999
Change-Id: Id3b2cd6c7e0923adfffb4e04c35ed2d6faca6704
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215921
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33802}
No customers have been identified.
Bug: chromium:1197965
Change-Id: Ia3063d0909c718ffb8e824225c8c60180551115a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214963
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33700}
This causes build failures in the Chromium fuzzers, so let's disable it
for now.
Bug: none
Change-Id: I0a076c0cd5cfb7d62383d733f3934f8b58f8ad34
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215040
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33693}
Each file is a SCTP packet (without any additional headers), all
extracted from a few Wireshark dumps that have been manually recorded.
Bug: webrtc:12614
Change-Id: I64bef0c563f1d83ae22735d702c8abafec6429b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214701
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33675}
This fuzzer explores the SCTP parsing, as well as the individual
chunks, as a successfully parsed packet will have its chunks iterated
over and formatted using ToString.
Bug: webrtc:12614
Change-Id: I88f703c5f79e4775a069b1d5439d413870f6a629
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214490
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33670}
This reverts commit c184047fef005b86a6dd76f03b0eb5ec01de3c5c.
Reason for revert: Breaks the WebRTC->Chromium roll:
ERROR Unresolved dependencies.
//third_party/webrtc/test/fuzzers:vp9_encoder_references_fuzzer(//build/toolchain/win:win_clang_x64)
needs //third_party/webrtc/modules/video_coding:mock_libvpx_interface(//build/toolchain/win:win_clang_x64)
We need to add tryjob to catch these. The fix is to make
//third_party/webrtc/modules/video_coding:mock_libvpx_interface
visible in built_with_chromium builds by moving the target
out of this "if" https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/modules/video_coding/BUILD.gn;l=615;drc=3889de1c4c7ae56ec742fb9ee0ad89657f638169.
Original change's description:
> Add fuzzer to validate libvpx vp9 encoder wrapper
>
> Fix simulcast svc controller to reuse dropped frame configuration,
> same as full svc and k-svc controllers do.
> This fuzzer reminded the issue was still there.
>
> Bug: webrtc:11999
> Change-Id: I74156bd743124723562e99deb48de5b5018a81d0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212281
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33568}
TBR=danilchap@webrtc.org,sprang@webrtc.org
Change-Id: I1676986308c6d37ff168467ff2099155e8895452
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11999
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212973
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33573}
Fix simulcast svc controller to reuse dropped frame configuration,
same as full svc and k-svc controllers do.
This fuzzer reminded the issue was still there.
Bug: webrtc:11999
Change-Id: I74156bd743124723562e99deb48de5b5018a81d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212281
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33568}
This reverts commit cd5127b11e04b7f49031b2412625c36e4a86c3da.
Reason for revert: Fuzzer build problems fixed.
Original change's description:
> Revert "Enable use of rtc::SystemTimeNanos() provided by Chromium"
>
> This reverts commit dfe19719e53abfd4d73722942445c5e1046b671b.
>
> Reason for revert: Breaks fuzzers in Chromium builds. See https://ci.chromium.org/ui/p/chromium/builders/try/linux-libfuzzer-asan-rel/685438/overview. I am reverting since this blocks the roll but I will be in touch for a fix.
>
> Original change's description:
> > Enable use of rtc::SystemTimeNanos() provided by Chromium
> >
> > This is the third CL out of three to enable overriding
> > of the function SystemTimeNanos() in rtc_base/system_time.cc
> >
> > When WebRTC is built as part of Chromium the rtc::SystemTimeNanos()
> > function provided by Chromium will be used. This is controlled
> > by the build argument rtc_exclude_system_time which directly
> > maps to the macro WEBRTC_EXCLUDE_SYSTEM_TIME.
> >
> > By doing this we are making sure that the WebRTC and Chromium
> > clocks are the same.
> >
> > Bug: chromium:516700
> > Change-Id: If7f749c4aadefb1cfc07ba4c7e3f45dc6c31118b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208223
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33337}
>
> TBR=kron@webrtc.org
>
> Bug: chromium:516700
> Change-Id: I9ecd1784a6c1cdac8bae07d34f7df20c62a21a95
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208740
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33340}
Bug: chromium:516700
Change-Id: I4cd68bac1cc4befdb46351f5d6fb2cf1ef5c3062
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208742
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33341}
This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a
Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which
affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5.
The original CL didn't attach the definition of the macro
NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have
to be related to //rtc_base anymore but to //rtc_base:threading).
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
Bug: webrtc:9987
Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33001}
This reverts commit 69241a93fb14f6527a26d5c94dde879013012d2a.
Reason for revert: Breaks WebRTC roll into Chromium.
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
TBR=mbonadei@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
No-Try: True
Bug: webrtc:9987
Change-Id: I1e36ad64cc60092f38d6886153a94f1a58339256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201840
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32986}
The "//rtc_base:rtc_base" build target has historically been one of the
biggest targets in the WebRTC build. Big targets are the main source of
circular dependencies and non-API types leakage.
This CL is a step forward into splitting "//rtc_base:rtc_base" into
smaller targets (as originally started in 2018).
The only non-automated changes are (like re-wiring the build system):
* The creation of //rtc_base/async_resolver.{h,cc} which allows to
break a circular dependency (is has been extracted from
//rtc_base/net_helpers.{h,cc}).
* The creation of //rtc_base/internal/default_socket_server.{h,cc} to
break another circular dependency.
Bug: webrtc:9987
Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32941}
