Left in target are just .cc files with .h files used externally.
Bug: webrtc:14775
Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41835}
describing video codecs with their parameters as static members of SdpVideoFormat:
static const SdpVideoFormat VP8();
static const SdpVideoFormat H264();
static const SdpVideoFormat VP9Profile0();
static const SdpVideoFormat VP9Profile1();
static const SdpVideoFormat VP9Profile2();
static const SdpVideoFormat VP9Profile3();
static const SdpVideoFormat AV1Profile0();
static const SdpVideoFormat AV1Profile1();
This removes the need to craft instances of these by hand.
BUG=webrtc:15703
Change-Id: I2171e08b48ec98f18424f53f3b5d6d148130532e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337441
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41833}
This reverts commit d99499abbae94793a02944a1f28f7015816447f5.
Reason for revert: Breaks downstream projects and I can also repro locally when running the rtc_unittest test target (it does however pass in isolation indicating test cleanup/setup needs to be fixed)
Original change's description:
> p2p: separate ICE tie breaker and foundation seed
>
> BUG=webrtc:14626
>
> Change-Id: I189a708192c9cef0b50c3fcbe798b30376d3b547
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338982
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41806}
Bug: webrtc:14626
Change-Id: If45f8a33395c562c9388b3d3748e8566efa87ecb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341081
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Christoffer Dewerin <jansson@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Owners-Override: Christoffer Dewerin <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41812}
The feature isn't in use by Google and has proven to contain security
issues. It's time to remove it.
Bug: b/324864439
Change-Id: I80344eb2f2060469d2d69a54dc4519fdd02ab4ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340324
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41808}
Goal is to make PeerConnectionInterface methods pure virtual.
This is a split of cl https://webrtc-review.googlesource.com/c/src/+/340143 in order to be able to fix Chromium test RTCPeerConnectionHandlerTest.OnRenegotiationNeeded
Bug: none
Change-Id: I5eac4d9a96c1b594c9e2b3505ef2466046065dc8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340481
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41798}
To ensure CreateVideoDecoder is only used as a fallback when Create is not implemented,
and thus make it safer to migrate VideoDecoderFactory implementations to Create.
Bug: webrtc:15791
Change-Id: Ifb15cf1d303348949ba51a3bb4c91b855a06627f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339841
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41777}
Move code from P2PTransportChannel to Candidate, where we set the
foundation value for remote prflx candidates.
Bug: none
Change-Id: I7dbcb85bca35dca7297136b0706092dd8d2b153c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339902
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41774}
until the code duplication can be removed which requires breaking
up the circular dependency.
BUG=webrtc:15847
Change-Id: Icc5f27dfcda26b1fcf16b19f79005d8b52fb6af3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339903
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41771}
Switch to type_name() for things like logging since `type()` will
change to returning an enumeration value.
The functional change that this has is that log statements and
Connection::ToString() (used for logging) will contain "host"
instead of "local" and "srflx" instead of "stun".
Bug: webrtc:15846
Change-Id: I35c50d026e4578a25d51765d59c6f2e01b850c94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41757}
Candidate::type() is currently how the name of the type is fetched,
but that getter returns a non-standard type name.
Instead, I'm adding a new getter, type_name(), will follow up with
updating dependent code that needs the string, to use type_name (and
adapt to potential dependency on "local" or "stun") and then switch
type() to be enum based.
Also adding a test file for Candidate with a couple of basic tests to
start with.
Bug: webrtc:15846
Change-Id: I9b78b2405a9f962a3c07eaa8e72a79854c6f5ceb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339660
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41740}
First commit in a series of commits to wire up the test output path utility that adds a random directory in the path, for problematic tests that run in concurrent execution environments.
Bug: webrtc:15833
Change-Id: I5e5b3940007be773d77dbbfc953efa810e4e3ea9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339522
Reviewed-by: Artem Titov <titovartem@webrtc.org>
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/main@{#41738}
This is a reland of commit 050ffefd854f8a57071992238723259e9ae0d85a
Original change's description:
> Extends WebRTC logs for software encoder fallback
>
> This CL extends logging related to HW->SW fallbacks on the encoder
> side in WebRTC. The goal is to make it easier to track down the
> different steps taken when setting up the video encoder and why/when
> HW encoding fails.
>
> Current logs are added on several lines which makes regexp searching
> difficult. This CL adds all related information on one line instead.
>
> Three new search tags are also added VSE (VideoStreamEncoder), VESFW
> (VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
>
> It has been verified that these added logs also show up in WebRTC
> logs in Meet.
>
> Logs from the GPU process are not included due to the sandboxed
> nature which makes it much more complex to add to the native
> WebRTC log. I think that these simple logs will provide value as is.
>
> Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
>
> Bug: b/322132132
> Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41733}
NOTRY=true
Bug: b/322132132
Change-Id: I25dd34b9ba59ea8502e47b4c89cd111430636e08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339680
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41736}
This reverts commit 050ffefd854f8a57071992238723259e9ae0d85a.
Reason for revert: Breaks downstream project.
Original change's description:
> Extends WebRTC logs for software encoder fallback
>
> This CL extends logging related to HW->SW fallbacks on the encoder
> side in WebRTC. The goal is to make it easier to track down the
> different steps taken when setting up the video encoder and why/when
> HW encoding fails.
>
> Current logs are added on several lines which makes regexp searching
> difficult. This CL adds all related information on one line instead.
>
> Three new search tags are also added VSE (VideoStreamEncoder), VESFW
> (VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
>
> It has been verified that these added logs also show up in WebRTC
> logs in Meet.
>
> Logs from the GPU process are not included due to the sandboxed
> nature which makes it much more complex to add to the native
> WebRTC log. I think that these simple logs will provide value as is.
>
> Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
>
> Bug: b/322132132
> Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41733}
Bug: b/322132132
Change-Id: I24d0a4e71a43ac192485f1af208563a51d919865
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339661
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41735}
This CL extends logging related to HW->SW fallbacks on the encoder
side in WebRTC. The goal is to make it easier to track down the
different steps taken when setting up the video encoder and why/when
HW encoding fails.
Current logs are added on several lines which makes regexp searching
difficult. This CL adds all related information on one line instead.
Three new search tags are also added VSE (VideoStreamEncoder), VESFW
(VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
It has been verified that these added logs also show up in WebRTC
logs in Meet.
Logs from the GPU process are not included due to the sandboxed
nature which makes it much more complex to add to the native
WebRTC log. I think that these simple logs will provide value as is.
Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
Bug: b/322132132
Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41733}
To reduce number calls to the CreateVideoDecoder
Bug: webrtc:15791
Change-Id: I5d6ecc2e5e68165d4e012b3ad7edb6eaa40e1913
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336420
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41706}
Rework transport_description_factory to only have non-DTLS mode for
testing, and rewrite tests accordingly.
Bug: webrtc:11066, chromium:804275
Change-Id: Ie7d477c4331c975e4e0a3034fbbb749ed9009446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336880
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41697}
Using the Api, BWE components are recreated and new settings can be
applied. Initially, the only configuration available is allowing BWE probes without media".
Note that BWE components are created when transport first becomes writable. So calling this method before a PeerConnection is connected is cheap and only changes configuration.
Integration test in https://webrtc-review.googlesource.com/c/src/+/337322
Bug: webrtc:14928
Change-Id: If2c848489bf94a1f7a5ebf90d2886d90c202c7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336240
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41687}
This completes the breakup of the rtc_p2p target.
Remaining cleanup is to delete the rtc_p2p target and make clients
depend on the base targets.
Bug: webrtc:15796
Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41683}
This is a reland of commit 59f3b35013a29f8c73a46fa6fd06aadc96aad892
Landing after taking out the Chrome usages.
Original change's description:
> Take out Fuchsia-only SDES-enabling parameters
>
> This does not remove all traces of SDES - we still need to delete
> the cricket::CryptoParams struct and all code that uses it.
>
> Bug: webrtc:11066, chromium:804275
> Change-Id: I811c8d40da7f4af714d53376f24cd53332a15945
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336780
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41634}
Bug: webrtc:11066, chromium:804275
Change-Id: I31414dfb6a0ecfa7b6fd91c68603cfd6146869d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337260
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41660}
This reverts commit 59f3b35013a29f8c73a46fa6fd06aadc96aad892.
Broke WebRTC into Chrome rolls:
https://chromium-review.googlesource.com/c/chromium/src/+/5248171?tab=checks
/../third_party/blink/renderer/modules/peerconnection/rtc_peer_connection_handler.cc:216:18: error: no member named 'enable_dtls_srtp' in 'webrtc::PeerConnectionInterface::RTCConfiguration'
216 | configuration->enable_dtls_srtp = dtls_srtp_key_agreement;
| ~~~~~~~~~~~~~ ^
Original change's description:
> Take out Fuchsia-only SDES-enabling parameters
>
> This does not remove all traces of SDES - we still need to delete
> the cricket::CryptoParams struct and all code that uses it.
>
> Bug: webrtc:11066, chromium:804275
> Change-Id: I811c8d40da7f4af714d53376f24cd53332a15945
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336780
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41634}
Bug: webrtc:11066, chromium:804275
Change-Id: I2c2114873091e0c662977a6ef5723e6447166a65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337181
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41643}
This does not remove all traces of SDES - we still need to delete
the cricket::CryptoParams struct and all code that uses it.
Bug: webrtc:11066, chromium:804275
Change-Id: I811c8d40da7f4af714d53376f24cd53332a15945
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336780
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41634}
Includes removing the duplicate MockTransformableAudioFrame definition
in test/ in favour of the existing one in api/test/
Bug: webrtc:15802
Change-Id: Ib5f86b8b2095dd4e580cd9ff0038134f8a43cd93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336340
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41622}
This CL does:
- Run IWYU on the relevant elements
- Make connection depend on port_interface, not port
- Make port_allocator depend only on port
- Move some constants from port.h into p2p_constants
This allows a dependency graph without ugly groups.
Bug: webrtc:15796
Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41618}
With this CL the migration to absl::optional<T> is complete.
Bug: webrtc:15164
Change-Id: I978d86833fbd39154ed9026206a02ea7496f8c4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335640
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41616}
With this CL, the only usage of RTCStatsMember within WebRTC is the
actual type alias declaration. It's not referenced anywhere anymore.
This allows us to deleting the type alias, but let's do that in a
standalone CL in case it gets reverted.
Bug: webrtc:15164
Change-Id: I766d07abb62b5ddd524859b8ed749394fc439e52
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335621
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41612}
This CL splits many of the source files in p2p:rtc_p2p into individual
compile targets.
One target - connection_and_port - was left with multiple source files
because it was too tangled to detangle at once.
Bug: webrtc:15796
Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41611}
This is a reland of commit 79ac694d9b70fa9cd7b6a0f00bbee5d7fbbe64de
Original change's description:
> [Stats] Make RTCStatsMember<T> a type alias for absl::optional<T>.
>
> The moment we've all been waiting for.
>
> Step 1: Add type alias (this CL).
> Step 2: Migrate all uses of RTCStatsMember<T> to absl::optional<T>.
> Step 3: Delete type alias.
>
> Bug: webrtc:15164
> Change-Id: I00a7202c0b684fb2c57fcad4f501bccc167f1fa3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/main@{#41593}
# Only unrelated bot failures
NOTRY=True
Bug: webrtc:15164
Change-Id: I0f1991409326679a260cb24907808eaa28385350
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335960
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41603}
