regression from
https://webrtc-review.googlesource.com/c/src/+/197813
which attempted to cast the unsupported content type with
a sctp protocol to a application/datachannel one.
BUG=chromium:1171965
Change-Id: I87c63da83b9f49d968e9b045bb1079f687ab226e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204481
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33100}
adds metrics for bundle usage, differentiating between
* BUNDLE is not negotiated and there is only a datachannel,
* BUNDLE is not negotiated and there is at most one m-line per media type,
for unified-plan
* BUNDLE is not negotiated and there are multiple m-lines per media type,
* BUNDLE is negotiated and there is only a datachannel,
* BUNDLE is negotiated but there is at most one m-line per media type,
* BUNDLE is negotiated and there are multiple m-lines per media type,
and for plan-b
* BUNDLE is negotiated
* BUNDLE is not negotiated
BUG=webrtc:12383
Change-Id: I41afc4b08fd97239f3b16a8638d9753c69b46d22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202254
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33090}
- Remove the last sigslot from JsepTransportController.
- Tested the potential test failure on chromium blink test by importing
this CL to chromium source.
Bug: webrtc:11943
Change-Id: I107d05606350aff655ca73a5cb640dff1a7036ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202741
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33085}
Functionality wise, there should be no change with this CL, aside
from updating tests to anticipate OnPacketReceived to handle the packet
asynchronously (as already was the case via BaseChannel).
This only removes the network->worker hop out of the BaseChannel
class into the WebRTC MediaChannel implementations. However, it updates
the interface contract between BaseChannel and MediaChannel to align
with how we want things to work down the line, i.e. avoid hopping to
the worker thread for every rtp packet.
The following steps will be to update the video and voice channel
classes to call Call::DeliverPacket on the network thread and only
handle unsignalled SSRCs on the worker (exception case).
Bug: webrtc:11993
Change-Id: If0540874444565dc93773aee89d862f3bfc9c502
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202242
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33040}
The extmap-allow-mixed SDP attribute signals that one- and two-byte RTP
header extensions can be mixed. In practice, this also means that WebRTC
will support two-byte RTP header extensions when this is signaled by
both peers.
Bug: webrtc:9985
Change-Id: I80a3f97bab162c7d9a5acf2cae07b977641c039d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197943
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33036}
- Replace few sigslot usages in jsep_transport_controller.
- There is still one sigslot usages in this file so keeping the inheritance
and that is the reason for not having a binary size gain in this CL.
- Remaining sigslot will be removed in a separate CL.
Bug: webrtc:11943
Change-Id: Idb8fa1090b037c48eeb62f54cffd3c485cebfcda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190146
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33034}
it no longer reports just offers.
BUG=chromium:857004
Change-Id: Idf35b6fa98f3ee6637aeef6b11553947fea3ee25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202249
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33024}
(and a subclass of QueuedTask in one place, where needed for move
semantics).
Bug: webrtc:11339
Change-Id: I109de41a8753f177db1bbb8d21b6744eb3ad2de0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201734
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33021}
all invokes, as well as BaseChannel constructor and destructor
should run on the same task queue which allow to use
simpler cancellation of pending task on BaseChannel destruction
Bug: webrtc:12339
Change-Id: I311b6de940cc24cf6bb5b49e1bbd132fea2439e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202032
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33009}
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}
For the case where an unknown header extension URI is attempted
to be modified by SetOfferedRtpHeaderExtensions, WebRTC emitted
INVALID_PARAMETER. Fix this by emitting UNSUPPORTED_PARAMETER.
Bug: chromium:1051821
Change-Id: I98b68e1e3a3f90f9cfa0d45833f46a307c246ad0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201733
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32983}
SetNegotiatedHeaderExtensions_w queued a task to update the list
of negotiated header extensions on the signal thread from the
worker thread, in belief that a later call to
GetNegotiatedHeaderExtensions() would happen on the WebRTC proxies,
leading to the update happening before the readout. In downstream
project, this is not always the case.
Fix this by synchronously updating the list of negotiated header
extensions.
Bug: chromium:1051821
Change-Id: I3266292e7508bb7a22a3f7d871e82c12f60cfc83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201728
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32977}
This version uses relative_packet_arrival_delay as the target metric.
Bug: none
Change-Id: Ie6eb575ce4d13fd005f026862892b14bd4fb1135
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201620
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32962}
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}
for style consistency. This check is already done outside the method.
BUG=None
Change-Id: Ie1366fa57417258a301b02503ad76f304f4279a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198040
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#32853}
and reorganise the parsing
Bug: None
Change-Id: I21f08297429a0cc0265da00daa681d934fc43d66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196643
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32843}
This should allow us to remove some SDP parsing in Chromium.
Bug: webrtc:12215
Change-Id: Ib85593d1c9226b29f2ec18617f945c76eca3b2d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197806
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32840}
Two audio channels going into the AudioSource::Sink can either be
down-mixed to mono or encoded as stereo. This change enables WebRTC
users (such as Chromium) to query the number of audio channels actually
encoded. That information can in turn be used to tailor the audio
processing to the number of channels actually encoded.
This change fixes webrtc:8133 from a WebRTC perspective and will be
followed up with the necessary Chromium changes.
Bug: webrtc:8133
Change-Id: I8e8a08292002919784c05a5aacb21707918809c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197426
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32836}
These tests create multiple transceivers, and attempt to renegotiate.
They serve to show where the limit is for adequate performance (arbitrarily
set as one second).
This version should pass on all platforms; it only tests up to 16 tracks.
Bug: webrtc:12176
Change-Id: I1561a56f6a392dbfa954319c538a9959c3a6f590
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193061
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32820}
Using CRYPTO_BUFFERs instead of legacy X509 objects offers memory and
security gains, and will provide binary size improvements as well once
the default list of built-in certificates can be removed; the code
dealing with them still depends on the X509 API.
Implemented by splitting openssl_identity and openssl_certificate
into BoringSSL and vanilla OpenSSL implementations.
No-Try: True
Bug: webrtc:11410
Change-Id: I86ddb361b94ad85b15ebb8743490de83632ca53f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196941
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32818}
Instead of doing a separate Invoke for each channel, this CL first
gathers a list of operations to be performed on the signaling thread,
then does a single Invoke on the worker thread (and nested Invoke
on the network thread) to update all channels at once.
This includes the methods:
* Enable
* SetLocalContent/SetRemoteContent
* RegisterRtpDemuxerSink
* UpdateRtpHeaderExtensionMap
Also, removed the need for a network thread Invoke in
IsReadyToSendMedia_w by moving ownership of was_ever_writable_ to the
worker thread.
Bug: webrtc:12266
Change-Id: I31e61fe0758aeb053b09db84f234deb58dfb3d05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194181
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32817}
This reverts commit 72f638a9a279e7abb5534fa66a0ade2cf18ec1a7.
Reason for revert: downstream build failures
Original change's description:
> Use CRYPTO_BUFFER APIs instead of X509 when building with BoringSSL.
>
> Using CRYPTO_BUFFERs instead of legacy X509 objects offers memory and
> security gains, and will provide binary size improvements as well once
> the default list of built-in certificates can be removed; the code
> dealing with them still depends on the X509 API.
>
> Implemented by splitting openssl_identity and openssl_certificate
> into BoringSSL and vanilla OpenSSL implementations.
>
> Bug: webrtc:11410
> Change-Id: Idc043462faac5e4ab1b75bedab2057197f80aba6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174120
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: David Benjamin <davidben@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32811}
TBR=deadbeef@webrtc.org,mbonadei@webrtc.org,davidben@webrtc.org,hta@webrtc.org
Change-Id: Ib5e55cb5798a2f3d25a4460f5311d2e650d3fa82
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11410
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196742
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32812}
Using CRYPTO_BUFFERs instead of legacy X509 objects offers memory and
security gains, and will provide binary size improvements as well once
the default list of built-in certificates can be removed; the code
dealing with them still depends on the X509 API.
Implemented by splitting openssl_identity and openssl_certificate
into BoringSSL and vanilla OpenSSL implementations.
Bug: webrtc:11410
Change-Id: Idc043462faac5e4ab1b75bedab2057197f80aba6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174120
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: David Benjamin <davidben@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32811}
otherwise this shows up in the logs as unhandled when it has been handled.
BUG=None
Change-Id: Ic081312a266d7a7ffff6220d2979cefa29a8591e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196652
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32810}
The a=rtcp:9 IN IP4 0.0.0.0 line is required by JSEP to be generated,
but is also required to be ignored. This reduces log spew.
Bug: None
Change-Id: I984060d9693b9df4c4cfdf2c5dea0ea620f4bc83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196641
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32798}
Example of current output in appr.tc:
https://paste.googleplex.com/4582802164023296
No-Try: True
Bug: None
Change-Id: I9b717b9c13e771e84682d9e3d3ee6b0920a85a44
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196526
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32796}
This moves the code for threadjumping to get the RTP transport
despite its thread guard from the main function to two functions
marked especially "ForTesting".
Bug: webrtc:12230
Change-Id: I4473ed38e6fdedb05e2fbc97c2521bc1993fdd1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196521
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32792}
This makes it thread-safe to access, but not necessarily to use.
Bug: webrtc:12230
Change-Id: I6b48d86dff24b162d382135abeaf560971fdf614
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196524
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32785}
guarded by a new field trial flag WebRTC-Debugging-RtpDump.
Packets have a RTP_DUMP postfix for easy grep-ing.
BUG=webrtc:10675
Change-Id: I73c0e0db47dca1079cd303c41a8b80fd7ae4a902
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196087
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32775}
Needed in order to return different codes for different failures
in initialization.
Sideswipe: Check TURN URL hostnames for illegal characters.
Bug: webrtc:12238
Change-Id: I1af3a37b9654b83b268304f7356049f9f3786b7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195541
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32710}
This CL also adds commentary to member variables that couldn't be guarded
because they're accessed from multiple threads.
Bug: webrtc:12230
Change-Id: I5193a7ef36ab25588c76ee6a1863de6a844be1dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195331
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32705}
