Commit Graph

357 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
1858995fa8 Reduce memory usage of vp9_encoder_references_fuzzer
Use cicular buffer instead of ever growing dynamic vector
That limits used memory and speed up fuzzing

Bug: chromium:1207177, chromium:1202535
Change-Id: Ia69ee7423f720942301b6d0b1a9c16a0cf1b3d55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218602
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34002}
2021-05-13 10:15:05 +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
87f7090fd9 Replace more instances of rtc::RefCountedObject with make_ref_counted.
This is essentially replacing `new rtc::RefCountedObject` with
`rtc::make_ref_counted` in many files. In a couple of places I
made minor tweaks to make things compile such as adding parenthesis
when they were missing.

Bug: webrtc:12701
Change-Id: I3828dbf3ee0eb0232f3a47067474484ac2f4aed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215973
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33852}
2021-04-27 17:01:59 +00:00
e8080f4a43 Fix uninitialized variable in vp9_encoder_references_fuzzer
Bug: chromium:1201537
Change-Id: Ic900340ebb632a40ed8c34a6e226e83b7a000203
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215962
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33819}
2021-04-23 13:02:19 +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
ce423ce12d Track last packet receive times in RtpVideoStreamReceiver instead of the PacketBuffer.
Bug: webrtc:12579
Change-Id: I4adb8c6ada913127b9e65d97ddce0dc71ec6ccee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214784
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33713}
2021-04-13 18:24:45 +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
4f88a9d1c3 Create a VideoFrameTrackingId RTP header extension.
Bug: webrtc:12594
Change-Id: I518b549b18143f4711728b4637a4689772474c45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212084
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/master@{#33567}
2021-03-25 17:25:18 +00:00
ca18809ee5 Move RtpFrameObject and EncodedFrame out of video_coding namespace.
Bug: webrtc:12579
Change-Id: Ib7ecd624eb5c54abb77fe08440a014aa1e963865
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33542}
2021-03-23 14:22:47 +00:00
6a6715042a Move RtpFrameReferenceFinder out of video_coding namespace.
Namespace used because of copy-pasting an old pattern, should never have been used in the first place. Removing it now to make followup refactoring prettier.

Bug: webrtc:12579
Change-Id: I00a80958401cfa368769dc0a1d8bbdd76aaa4ef5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212603
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33536}
2021-03-23 08:48:37 +00:00
7c7885c016 Remove NTP timestamp from PacketBuffer::Packet.
Bug: webrtc:12579
Change-Id: I64ca0ddb6f5c20bef5e9503955e0e4b4c484a1e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211662
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33448}
2021-03-12 15:19:35 +00:00
f00cd533ae Do more actions on SDP fuzzing.
This will wait for completion of the setRemoteDescription() call,
and if the setRemoteDescription() is successful, it will do a
setLocalDescription() (default description).

Bug: none
Change-Id: Id1cb6b1ecbdc90d4f2c5b46a7f8e92b7491ff401
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210682
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33400}
2021-03-08 12:18:10 +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
07d83c8a9a Modified STUN verification functions
The new verification makes verification a function on a message.
It also stores the password used in the request message, so that
it is easily accessible when verifying the response.

Bug: chromium:1177125
Change-Id: I505df4b54214643a28a6b292c4e2262b9d97b097
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209060
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33366}
2021-03-02 10:34:17 +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
9aa9b8dbbe Prepare to replace VideoLayerFrameId with int64_t.
Bug: webrtc:12206
Change-Id: I10bfdefbc95a79e0595956c1a0e688051da6d2b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207180
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33265}
2021-02-15 14:42:02 +00:00
f109193fba Remove VideoLayerFrameId::spatial_layer, use EncodedImage::SpatialIndex instead.
Next step is to replace VideoLayerFrameId with int64_t.

Bug: webrtc:12206
Change-Id: I414f491e383acf7f8efd97f7bf93dc55a5194fbf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206804
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33245}
2021-02-12 11:16:23 +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
a782022d6c Use rtc::CopyOnWriteBuffer::MutableData instead of data in fuzzers
Bug: webrtc:12334
Change-Id: I3df42998f5cf5c3b09ad3f6253cab34170d725bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201380
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32947}
2021-01-12 12:36:43 +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
6afa794b6e Delete deprecated H264BitstreamParser methods
Bug: webrtc:10439
Change-Id: I1513907f03f9adfcf5657298e69d60519af764ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198121
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32934}
2021-01-11 09:31:54 +00:00
62a9a32937 In Av1 packetizer set marker bit with respect of end_of_picture flag
Bug: webrtc:12167
Change-Id: If14fdd7144951c7aa7e48efd390637dd66201bf7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192791
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32612}
2020-11-16 11:08:48 +00:00
6556ed2402 Add experimental extension RtpVideoLayersAllocation
The extension is suggested to be used for signaling per target bitrate, resolution
and frame rate to a SFU to allow a SFU to know what video layers a client is currently targeting.
It is hoped to replace the current Target bitrate RTCP XR message currently used only for screen share.

Bug: webrtc:12000
Change-Id: Id7b55e7ddaf6304e31839fd0482b096e1dbe8925
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185980
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32313}
2020-10-05 13:38:13 +00:00
df2a4654a0 Improve neteq_rtp_fuzzer
This change lets the fuzzer modify the first few bytes of the RTP
payload. One of the benefits is that it can cover the RED header
splitter functionality.

The CL also fixes an issue found while running the fuzzer locally.

Bug: webrtc:11640
Change-Id: I7ca73676440897a14a0aaca796f70d381e016575
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185819
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32242}
2020-09-29 20:24:07 +00:00
d381eede92 Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h
We can then finally delete the top-level common_types.h, and the
corresponding build target webrtc_common.

Bug: webrtc:7660
Change-Id: I1c1096541477586d90774c7a3405b9d36edec14a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182800
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32044}
2020-09-07 08:37:14 +00:00
2b068ce1b8 Reland "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
This reverts commit a4f23ad0ce4382e3a11bc6a8c1f9f6183e722fd8.

Reason for revert: Downstream fix landed.

TBR=mflodman@webrtc.org

Original change's description:
> Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
>
> This reverts commit acb9d8365a5f9eb1e2a9e9902690d62dab1e5759.
>
> Reason for revert: Break downstream stuff.
>
> Original change's description:
> > Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory.
> >
> > Bug: webrtc:9106
> > Change-Id: I85712f3ab6a734d3fad7819491d3b8e3388b47e7
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180342
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31834}
>
> TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org
>
> Change-Id: I6cfdb85a154a78135839f84edf5f69673d5ab715
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9106
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180807
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31835}

TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org

# Not skipping CQ checks because this is a reland.

Bug: webrtc:9106
Change-Id: I03b3e68532107bec37bcc6e47a5489c84fe91ef9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180808
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31866}
2020-08-06 11:50:08 +00:00
a4f23ad0ce Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
This reverts commit acb9d8365a5f9eb1e2a9e9902690d62dab1e5759.

Reason for revert: Break downstream stuff.

Original change's description:
> Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory.
> 
> Bug: webrtc:9106
> Change-Id: I85712f3ab6a734d3fad7819491d3b8e3388b47e7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180342
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31834}

TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org

Change-Id: I6cfdb85a154a78135839f84edf5f69673d5ab715
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9106
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180807
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31835}
2020-08-03 15:45:41 +00:00
acb9d8365a Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory.
Bug: webrtc:9106
Change-Id: I85712f3ab6a734d3fad7819491d3b8e3388b47e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180342
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31834}
2020-08-03 14:10:37 +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
9465978a3b Remove framemarking RTP extension.
BUG=webrtc:11637

Change-Id: I47f8e22473429c9762956444e27cfbafb201b208
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176442
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31522}
2020-06-15 11:18:00 +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
f25761d798 Remove dependency from RtpRtcp on the Module interface.
The 'Module' part of the implementation must not be
called via the RtpRtcp interface, but is rather a part of
the contract with ProcessThread. That in turn is an
implementation detail for how timers are currently implemented
in the default implementation.

Along the way I'm deprecating away the factory function which
was inside the interface and tied it to one specific implementation.
Instead, I'm moving that to the implementation itself and down the
line, we don't have to go through it if we just want to create an
instance of the class.

The key change is in rtp_rtcp.h and the new rtp_rtcp_interface.h
header file (things moved from rtp_rtcp.h), the rest falls from that.

Change-Id: I294f13e947b9e3e4e649400ee94a11a81e8071ce
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176419
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31440}
2020-06-04 08:11:21 +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
3580706684 Add a RunLoop to RtpReplayer to fix fuzzers
Bug: chromium:1080852
Change-Id: Ia02511cde09994deee222e4f1267d5265d0364ca
Tbr: mbonadei@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174756
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31196}
2020-05-09 06:45:14 +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
4c3a7dbe14 Remove RtpVideoHeader::discardable flag.
Calculate it when used instead

Bug: webrtc:11358
Change-Id: Ib79a4ce5e48a1a5244925471c005f96c5ec5dfd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173702
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31109}
2020-04-20 10:25:43 +00:00
ec9fc2208e Delete generic frame descriptor v1 trait and enum value
Bug: webrtc:11358
Change-Id: I272a45881f8ef9963b502c6d17edc97e7d9fbc8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173582
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@{#31089}
2020-04-16 17:29:18 +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