Commit Graph

157 Commits

Author SHA1 Message Date
3d2a3355e3 dcsctp: Add socket fuzzer
Bug: webrtc:12614
Change-Id: I43659e96fbd44a10b3e8d690602afa4673df1228
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218501
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34008}
2021-05-14 06:41:10 +00:00
fe2063ebc7 Remove REMB throttling funcionality from PacketRouter
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}
2021-05-12 11:24:58 +00:00
e7b752b221 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.

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}
2021-04-21 14:29:04 +00:00
c8cf0a6080 Remove MDNS message implementation
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}
2021-04-12 22:24:56 +00:00
3928e8fdb1 dcsctp: Disable packet fuzzers
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}
2021-04-12 14:49:09 +00:00
50fc1dfbcc dcsctp: Add SCTP packet corpus
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}
2021-04-11 18:25:08 +00:00
9410217413 dcsctp: Add SCTP packet fuzzer
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}
2021-04-09 12:23:42 +00:00
6e6411c099 Revert "Add fuzzer to validate libvpx vp9 encoder wrapper"
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}
2021-03-26 11:17:00 +00:00
c184047fef 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}
2021-03-25 18:52:38 +00:00
82a94125a4 Reland "Add a fuzzer test that tries to connect a PeerConnection."
This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38.

Reason for revert: Added Chromium compile guards

Original change's description:
> Revert "Add a fuzzer test that tries to connect a PeerConnection."
>
> This reverts commit c67b77eee4b08c05638a219723a9141a65015da4.
>
> Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll.
>
> Original change's description:
> > Add a fuzzer test that tries to connect a PeerConnection.
> >
> > Bug: none
> > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33369}
>
> NOPRESUBMIT=true
>
> Bug: none
> Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33380}

Bug: none
Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 08:58:09 +00:00
ae44fde188 Revert "Add a fuzzer test that tries to connect a PeerConnection."
This reverts commit c67b77eee4b08c05638a219723a9141a65015da4.

Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll.

Original change's description:
> Add a fuzzer test that tries to connect a PeerConnection.
>
> Bug: none
> Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33369}

NOPRESUBMIT=true

Bug: none
Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33380}
2021-03-04 09:42:34 +00:00
c67b77eee4 Add a fuzzer test that tries to connect a PeerConnection.
Bug: none
Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33369}
2021-03-02 19:28:23 +00:00
bb52bdf095 Reland "Enable use of rtc::SystemTimeNanos() provided by Chromium"
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}
2021-02-25 10:48:55 +00:00
e5f4c6b8d2 Reland "Refactor rtc_base build targets."
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}
2021-01-15 17:00:05 +00:00
7acc2d9fe3 Revert "Refactor rtc_base build targets."
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}
2021-01-14 21:27:38 +00:00
69241a93fb 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}
2021-01-11 18:32:30 +00:00
c1b271264a Delete RtcpDemuxer as unused
Bug: None
Change-Id: I17b30af3fef6c165bf951cb58eef11cc9c37aa39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178396
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31676}
2020-07-08 14:36:20 +00:00
96115cfcdd Add absl_deps to webrtc_fuzzer_test.
Bug: chromium:1046390
Change-Id: I531511dce156a10174c9ed80ccb2d5cd75ec33b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177900
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31553}
2020-06-24 08:22:30 +00:00
2dcf348011 Use absl_deps in order to preapre to the Abseil component build release.
Bug: webrtc:1046390
Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-08 12:59:40 +00:00
fe6a353ce4 fuzzers: fix isax typo
TBR=saza@webrtc.org
BUG=none

Change-Id: If565fbcca92f162b9483eb6abeaf3c374998c2df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176123
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31355}
2020-05-26 12:51:28 +00:00
b0bd0708d6 Surface ResidualEchoDetector creation to API
This allows users to inject the residual echo detector, as a step toward making it an optional part of compilation.

Bug: webrtc:11292, webrtc:11539
Change-Id: I7fcc8dbaced67a82851cd6cdcbc115eb01c21fcf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174040
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31222}
2020-05-12 10:56:18 +00:00
cc73ed3e70 APM: Add build flag to allow building WebRTC without APM
This CL adds a build flag to allow building the non-test parts
of WebRTC without the audio processing module.
The CL also ensures that the WebRTC code correctly handles
the case when no APM is available.

Bug: webrtc:5298
Change-Id: I5c8b5d1f7115e5cce2af4c2b5ff701fa1c54e49e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171509
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31133}
2020-04-26 23:06:44 +00:00
fea8b94591 Reland "APM: Remove the usage of AudioFrame in the AudioProcessing interface"
This is a reland of 12e2d4ddb235da6ec7a5c1c3a83ac33d394920b0

Original change's description:
> APM: Remove the usage of AudioFrame in the AudioProcessing interface
> 
> This CL removes the AudioFrame-based APIs from the AudioProcessing
> interface.
> 
> Bug: webrtc:5298
> Change-Id: Iab470b26b10e06dcf29c543851ae0085bc5b66f0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172939
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31016}

Bug: webrtc:5298
Change-Id: I70e6d59afc3716ee6109d8b9dc384abc71c93624
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173476
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31066}
2020-04-14 14:11:06 +00:00
7e60483915 Revert "APM: Remove the usage of AudioFrame in the AudioProcessing interface"
This reverts commit 12e2d4ddb235da6ec7a5c1c3a83ac33d394920b0.

Reason for revert: Speculative revert: breaks downstream project

Original change's description:
> APM: Remove the usage of AudioFrame in the AudioProcessing interface
> 
> This CL removes the AudioFrame-based APIs from the AudioProcessing
> interface.
> 
> Bug: webrtc:5298
> Change-Id: Iab470b26b10e06dcf29c543851ae0085bc5b66f0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172939
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31016}

TBR=saza@webrtc.org,peah@webrtc.org

Change-Id: I82729b54c74cf1362332a28a96f598d6747b53ff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:5298
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173091
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31022}
2020-04-07 19:37:32 +00:00
12e2d4ddb2 APM: Remove the usage of AudioFrame in the AudioProcessing interface
This CL removes the AudioFrame-based APIs from the AudioProcessing
interface.

Bug: webrtc:5298
Change-Id: Iab470b26b10e06dcf29c543851ae0085bc5b66f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172939
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31016}
2020-04-07 13:40:58 +00:00
f87536c9de Reland "Reland "Refactors UlpFec and FlexFec to use a common interface.""
This is a reland of 49734dc0faa69616a58a1a95c7fc61a4610793cf

Patchset 2 contains a fix for the fuzzer set up. Since we now parse
an RtpPacket out of the fuzzer data, the header needs to be correct,
otherwise we fail before even reaching the FEC code that we actually
want to test.

Bug: webrtc:11340, chromium:1052323, chromium:1055974
TBR=stefan@webrtc.org

Original change's description:
> Reland "Refactors UlpFec and FlexFec to use a common interface."
>
> This is a reland of 11af1d7444fd7438766b7bc52cbd64752d72e32e
>
> Original change's description:
> > Refactors UlpFec and FlexFec to use a common interface.
> >
> > The new VideoFecGenerator is now injected into RtpSenderVideo,
> > and generalizes the usage.
> > This also prepares for being able to genera FEC in the RTP egress
> > module.
> >
> > Bug: webrtc:11340
> > Change-Id: I8aa873129b2fb4131eb3399ee88f6ea2747155a3
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168347
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30515}
>
> Bug: webrtc:11340, chromium:1052323
> Change-Id: Id646047365f1c46cca9e6f3e8eefa5151207b4a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168608
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30593}

Bug: webrtc:11340, chromium:1052323
Change-Id: Ib8925f44e2edfcfeadc95c845c3bfc23822604ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169222
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30724}
2020-03-09 13:41:35 +00:00
729310aa18 iSAC fixed|float encoder fuzzers
Bug: webrtc:11388
Change-Id: I5910492ef9471aa193aa50ef5e14b4b66cb6542a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169365
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30635}
2020-02-27 18:26:05 +00:00
02b76bd40b Opus Encoder fuzzer: separate target for FuzzAudioEncoder
Move FuzzAudioEncoder to a separate target to make it available for
other encoders.

Bug: webrtc:11388
Change-Id: I8b9a0f810791880eedb129b55eb33f154790e48f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169364
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30634}
2020-02-27 16:13:15 +00:00
c310889ec7 Revert "Reland "Refactors UlpFec and FlexFec to use a common interface.""
This reverts commit 49734dc0faa69616a58a1a95c7fc61a4610793cf.

Reason for revert: Still something wrong with ulpfec fuzzer setup.

Original change's description:
> Reland "Refactors UlpFec and FlexFec to use a common interface."
> 
> This is a reland of 11af1d7444fd7438766b7bc52cbd64752d72e32e
> 
> Original change's description:
> > Refactors UlpFec and FlexFec to use a common interface.
> >
> > The new VideoFecGenerator is now injected into RtpSenderVideo,
> > and generalizes the usage.
> > This also prepares for being able to genera FEC in the RTP egress
> > module.
> >
> > Bug: webrtc:11340
> > Change-Id: I8aa873129b2fb4131eb3399ee88f6ea2747155a3
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168347
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30515}
> 
> Bug: webrtc:11340, chromium:1052323
> Change-Id: Id646047365f1c46cca9e6f3e8eefa5151207b4a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168608
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30593}

TBR=sprang@webrtc.org,stefan@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11340, chromium:1052323
Change-Id: I920ce0a48a08768d7a98a563e2b66bd6eb8602b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169121
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30616}
2020-02-26 09:37:31 +00:00
49734dc0fa Reland "Refactors UlpFec and FlexFec to use a common interface."
This is a reland of 11af1d7444fd7438766b7bc52cbd64752d72e32e

Original change's description:
> Refactors UlpFec and FlexFec to use a common interface.
>
> The new VideoFecGenerator is now injected into RtpSenderVideo,
> and generalizes the usage.
> This also prepares for being able to genera FEC in the RTP egress
> module.
>
> Bug: webrtc:11340
> Change-Id: I8aa873129b2fb4131eb3399ee88f6ea2747155a3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168347
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30515}

Bug: webrtc:11340, chromium:1052323
Change-Id: Id646047365f1c46cca9e6f3e8eefa5151207b4a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168608
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30593}
2020-02-24 14:20:27 +00:00
cb4d380ba5 Revert "Refactors UlpFec and FlexFec to use a common interface."
This reverts commit 11af1d7444fd7438766b7bc52cbd64752d72e32e.

Reason for revert: Possible crash

Original change's description:
> Refactors UlpFec and FlexFec to use a common interface.
> 
> The new VideoFecGenerator is now injected into RtpSenderVideo,
> and generalizes the usage.
> This also prepares for being able to genera FEC in the RTP egress
> module.
> 
> Bug: webrtc:11340
> Change-Id: I8aa873129b2fb4131eb3399ee88f6ea2747155a3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168347
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30515}

TBR=brandtr@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org

Change-Id: Iddf112d801621c8a4370b853cee3fa42bf2c7fba
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168603
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30524}
2020-02-14 13:19:07 +00:00
11af1d7444 Refactors UlpFec and FlexFec to use a common interface.
The new VideoFecGenerator is now injected into RtpSenderVideo,
and generalizes the usage.
This also prepares for being able to genera FEC in the RTP egress
module.

Bug: webrtc:11340
Change-Id: I8aa873129b2fb4131eb3399ee88f6ea2747155a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168347
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30515}
2020-02-13 13:21:19 +00:00
190539717b Remove unused NextFrame function from FrameBuffer.
Also updated FrameBuffer unittests to use the GlobalSimulatedTimeController.

Bug: webrtc:7408, webrtc:9378
Change-Id: I8ade27492f66cdd8950b38f5f4a268714dbc35fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164536
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30422}
2020-01-30 12:54:08 +00:00
ccbe95fd8a Reformat GN files.
`gn format` recently [1] changed its formatting behavior
for deps, source, and a few other elements when they
are assigned (with =) single-element lists to be consistent
with the formatting of updates (with +=) with single-element.

Now that we've rolled in a GN binary with the change,
reformat all files so that people don't get presubmit
warnings due to this.

CL generated with:
$ git ls-files | grep BUILD.gn | xargs gn format
$ gn format build_overrides/build.gni
$ gn format build_overrides/gtest.gni
$ gn format modules/audio_coding/audio_coding.gni
$ gn format webrtc.gni
$ gn format .gn

Plus a few manual changes to add exceptions for
"public_deps" (after changing these lines the presubmit
started to complain).

[1] - https://gn-review.googlesource.com/c/gn/+/6860

Bug: webrtc:11302
Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30334}
2020-01-21 12:13:11 +00:00
d06588a758 Change Av1 depacketizer to implement VideoRtpDepacketizer interface
Bug: webrtc:11152
Change-Id: I322115263f60439bee36277157a0acef9bd28e3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165343
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30260}
2020-01-15 10:16:03 +00:00
7d43801a07 Delete RtpGenericDepacketizer as no longer used
Bug: webrtc:11152
Change-Id: I275765e1aa013d8188d43e2911e8ab022563d1d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165394
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30234}
2020-01-13 13:45:37 +00:00
5c35f2fb1b Delete RtpDepacketizerVp9 in favor of VideoRtpDepacketizerVp9
Bug: webrtc:11152
Change-Id: Ic50f2dc49ca420b3406d4dea11ed20328aa59136
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165382
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30195}
2020-01-09 13:07:44 +00:00
26e1b7ac01 Delete RtpDepacketizerVp8 in favor of VideoRtpDepacketizerVp8
Bug: webrtc:11152
Change-Id: I1a6225701ecd6f7a34c946d7296f0ab0cbb5eaef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165342
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30190}
2020-01-09 12:10:19 +00:00
242a9e0ffe Fuzz RtpPacketizerAv1
Bug: webrtc:11042
Change-Id: Id44699395f6dee9cb3bde84c936573b65ad0d848
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161009
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30007}
2019-12-04 13:52:51 +00:00
a9ad36f322 Fix aec3_fuzzer chromium build config.
Dependencies need to use relative paths in order to work in Chromium,
see [1].

[1] - https://ci.chromium.org/p/chromium/builders/try/linux-libfuzzer-asan-rel/334174

TBR: saza@webrtc.org
Bug: None
Change-Id: I50c401e5983fbb501d1da2ad909198261a8cb940
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161300
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30000}
2019-12-04 10:45:52 +00:00
b0db98cf06 Fuzz AEC3
This fuzzer fuzzes AEC3 with the default configuration and variable sample rates and channel counts.

Bug: None
Change-Id: I0d178a320b75fc4cc389657fa2b99931f359b517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160646
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29967}
2019-12-02 09:16:51 +00:00
5314b13a8d Fix undefined-shift in RtpDepacketizerAv1::AssembleFrame
Bug: chromium:1028348
Change-Id: I824e84138acbf4e73fc21ee8248e29e5cc7a0ba0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160643
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29945}
2019-11-28 11:27:33 +00:00
429d8fe28b Add fuzzer test for RtpDepacketizerAv1::AssembleFrame function
Bug: webrtc:11042
Change-Id: If5b7e0d81fd8c6590823ecab8f3909ed6c824f06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160016
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29893}
2019-11-25 10:45:38 +00:00
51868f52c6 Revert "Reland "Make webrtc_fuzzer_main depend on webrtc_component in Chromium.""
This reverts commit 8994c8bab315fa34b75a8e79b78bb99c86f69966.

Reason for revert: While RTC_EXPORTS are needed, this is still not
enough, I will try another approach, similar to what we do for
rtc_base/logging.{cc,h}.

Original change's description:
> Reland "Make webrtc_fuzzer_main depend on webrtc_component in Chromium."
> 
> This is a reland of 2148e9a931ea1a8a2ac0bfffd56e12370f8bf18c
> 
> Original change's description:
> > Make webrtc_fuzzer_main depend on webrtc_component in Chromium.
> >
> > This is needed in order to land [1] and restrict visibility of some
> > //third_party/webrtc_overrides targets.
> >
> > [1] - https://chromium-review.googlesource.com/c/chromium/src/+/1930801
> >
> > Bug: chromium:896154
> > Change-Id: Ie71c44ee9a0203a85d77a1199acdcb8581dfb71b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160308
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29875}
> 
> No-Try: True
> No-Tree-Checks: true
> TBR: kwiberg@webrtc.org
> Bug: chromium:896154
> Change-Id: I157bd4f90528a38ac16f17dd17af2f255dbd5ec9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160401
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29888}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

Change-Id: If969618e3f0a0cd70204128f1e8a2b06cf407b6e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:896154
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160402
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29889}
2019-11-23 15:10:47 +00:00
8994c8bab3 Reland "Make webrtc_fuzzer_main depend on webrtc_component in Chromium."
This is a reland of 2148e9a931ea1a8a2ac0bfffd56e12370f8bf18c

Original change's description:
> Make webrtc_fuzzer_main depend on webrtc_component in Chromium.
>
> This is needed in order to land [1] and restrict visibility of some
> //third_party/webrtc_overrides targets.
>
> [1] - https://chromium-review.googlesource.com/c/chromium/src/+/1930801
>
> Bug: chromium:896154
> Change-Id: Ie71c44ee9a0203a85d77a1199acdcb8581dfb71b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160308
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29875}

No-Try: True
No-Tree-Checks: true
TBR: kwiberg@webrtc.org
Bug: chromium:896154
Change-Id: I157bd4f90528a38ac16f17dd17af2f255dbd5ec9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160401
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29888}
2019-11-23 11:22:30 +00:00
3af0cd8de2 Revert "Make webrtc_fuzzer_main depend on webrtc_component in Chromium."
This reverts commit 2148e9a931ea1a8a2ac0bfffd56e12370f8bf18c.

Reason for revert: Breaks linux-libfuzzer-asan-rel,
https://ci.chromium.org/p/chromium/builders/try/linux-libfuzzer-asan-rel/326226. I will export symbols in this CL when relanding.

Original change's description:
> Make webrtc_fuzzer_main depend on webrtc_component in Chromium.
> 
> This is needed in order to land [1] and restrict visibility of some
> //third_party/webrtc_overrides targets.
> 
> [1] - https://chromium-review.googlesource.com/c/chromium/src/+/1930801
> 
> Bug: chromium:896154
> Change-Id: Ie71c44ee9a0203a85d77a1199acdcb8581dfb71b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160308
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29875}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

Change-Id: I1000e90e687d01c29a9ec4a3c8ded646b97fcaab
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:896154
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160400
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29885}
2019-11-23 08:41:57 +00:00
2148e9a931 Make webrtc_fuzzer_main depend on webrtc_component in Chromium.
This is needed in order to land [1] and restrict visibility of some
//third_party/webrtc_overrides targets.

[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1930801

Bug: chromium:896154
Change-Id: Ie71c44ee9a0203a85d77a1199acdcb8581dfb71b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160308
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29875}
2019-11-22 14:31:00 +00:00
56d945233d Move stun.h to api/.
We now have two downstream users of stun.h, so it appears to be
generally usable. I put this in a new dir networking/, but I'm open to
suggestions here (maybe some things in api/ should move in there).

I checked what our downstream users are actually using, and it's

cricket::ComputeStunCredentialHash
cricket::<constants>
cricket::TurnMessage
cricket::GetStunErrorResponseType
cricket::StunAttribute::CreateAddress
cricket::StunErrorCodeAttribute
cricket::StunByteStringAttribute
StunAttribute::CreateUnknownAttributes
cricket::TurnErrorType
cricket::StunMessage

I reckoned that was pretty much everything in stun.h, so I didn't
bother splitting it up. They don't use every function and constant
in there, but all _types_ of functions and constants, so for the
sake of coherence I don't think it makes sense to split it.

There's some old stuff in there like GTURN which could arguably
be split out, but it should likely go away soon anyway, so I don't
think it's worth the effort.

Steps:
1) land this
2) update downstream to point to the new header and target
3) remove p2p/base:stun_types.

Bug: webrtc:11091
Change-Id: I1f05bf06055475d25601197ec6fefb8d3b55e8e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159923
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29822}
2019-11-18 16:11:27 +00:00
ccf12c6e97 Reland "Add AV1 RtpDepacketizer class"
This is a reland of 49470c2ac460ed8cce250942e8525c5f14e32778
Tentative reland to rule-out bot flakiness.

Original change's description:
> Add AV1 RtpDepacketizer class
>
> Implement Parse function that extracts is_first_packet_in_frame,
> is_last_packet_in_frame, and frame_type fields.
>
> Bug: webrtc:11042
> Change-Id: I9360ea52ef274281b5c5e4c31955100b92155bfe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159180
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29814}

TBR=saza@webrtc.org,philipel@webrtc.org

Bug: webrtc:11042
Change-Id: Ibd672ce685bcab86960500740465539ed70fcdf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159941
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29819}
2019-11-18 15:23:08 +00:00
9f99175710 Revert "Add AV1 RtpDepacketizer class"
This reverts commit 49470c2ac460ed8cce250942e8525c5f14e32778.

Reason for revert: Seems to trigger linker error on iOS64. See:
https://ci.chromium.org/p/webrtc/builders/ci/iOS64%20Debug/17733

Original change's description:
> Add AV1 RtpDepacketizer class
> 
> Implement Parse function that extracts is_first_packet_in_frame,
> is_last_packet_in_frame, and frame_type fields.
> 
> Bug: webrtc:11042
> Change-Id: I9360ea52ef274281b5c5e4c31955100b92155bfe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159180
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29814}

TBR=danilchap@webrtc.org,saza@webrtc.org,philipel@webrtc.org

Change-Id: I2eb5994d8e31e12d6cb6e9f792b691ed10d9df81
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11042
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159940
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29815}
2019-11-18 12:14:56 +00:00