- add DTLS1.3 ciphers (without KeyType)
- remove code in dtls_transport.cc that tries to parse DTLS packet
- cleanup some test
- start on test for packet loss during dtls handshake (more to come!)
After this patch is submitted, it is possible
to set max version = dtls1.3 and it will active
but DON'T do it yet.
BUG=webrtc:383141571
Change-Id: I6f9a120c53415ccee7a560ea83bd0c2636702997
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371300
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43595}
This allows us to verify consistency of codec lists in more places.
Bug: webrtc:360058654
Change-Id: Ibd0d10579c4b8058031db0df458e8fc9e2181152
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371921
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43594}
Lists of codecs have a lot of cross references (RTX/APT and the like).
We should introduce functionality to verify that those linkages are correct
before modifying the handling of these.
This CL introduces the CodecList class, which can be extended to do
that verification. It is used by pc/media_session.cc, but inter-module
APIs are not changed in this version (they will be later).
Bug: webrtc:360058654
Change-Id: Ifd6313d0289cfa090e51ac28bc775265d18fe6f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371600
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43582}
In particular that avoids lifetime issues with the field trials passed into peerconnection, as now PC takes field trials object by unique_ptr and thus fully manages its lifetime.
Bug: webrtc:42220378
Change-Id: Ia863e9703b5c76ae1866d0ff995b83286c0b947e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371480
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43576}
This allows other tests using RTC stats to get pretty printing as well.
Bug: webrtc:381524905
Change-Id: Ib1eb9e1dad36b89e5b1c2ec687fcfeb308f82939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370761
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43575}
The negotiation of encrypted header extensions has already been enabled in Chromium, https://chromium-review.googlesource.com/c/chromium/src/+/5933829. Hence, it make sense to enable the encryption of header extensions by default also in webRTC environment so that all the tests run by taking this into considiration when new changes are made.
Bug: webrtc:358039777
Change-Id: I141fac01b0eb0f2ce5a0a365736f0dcf9f21ddcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43573}
This reverts commit e046787a5a80a9d292b3aec7e946644e025a2b95.
Reason for revert: Revised codec matching to fix issue.
Changes also back out some changes that should not have been
included (using PayloadTypePicker for codec list merging).
Original change's description:
> Revert "Use PayloadTypePicker for video PT assignment"
>
> This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3.
>
> Reason for revert: Broke internal tests.
>
> Original change's description:
> > Use PayloadTypePicker for video PT assignment
> >
> > This includes changes that change the order of codecs.
> > It is preparatory to doing late assignment of video PTs.
> >
> > Bug: webrtc:360058654
> > Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#43489}
>
> Bug: webrtc:360058654
> Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43490}
Bug: webrtc:360058654
Change-Id: I66b3b6bd657c66f8860c5e67a504266d7707f48d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370380
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43554}
https://webrtc.googlesource.com/src/+/7738bc23ed7fee0d4856bdfe7b88985865829441
switched from using sizeof(uint32_t) to SRTP_SRCTP_INDEX_LEN.
It turned out that this is not always defined.
This patch defines it to 4.
BUG=webrtc:42222036
Change-Id: Ice3d24a6300d19bc2f573469aadd6474ace1b147
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43548}
dtls_transport will when detecting a new fingerprint
(e.g by usage of pranswer) signal DtlsTransportState::kNew.
When this happen, the dtls crypto state is lost, and
sctp should reconnect, srtp does this automatically
in current code base.
The existing behavior in dcsctp is that it will detect
peer sending an init, and reconnect. But any messages sent
between the dtls restart and the message arriving from the
peer will be lost.
This patch changes so that this case is gracefully handled by
a) letting dcsctp_transport listen to dtls state
this is big part of patch and involves changing the type of
the underlying dtransport from rtc::PacketTransportInternal to cricket::DtlsTransportInternal. If requested, I can put this
into a separate patch...
b) if a dtls restart happens, delete and restart socket.
Testcase that fails before patch and works after is attached.
Bonus: And include-what-you-use on patch
Bug: b/375327137
Change-Id: Ib78488ae75fd8aeb50d121adf464a33dabbf95e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367202
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43546}
This reverts commit 7738bc23ed7fee0d4856bdfe7b88985865829441.
Reason for revert: Some downstream projects are still using an older version of libsrtp
Original change's description:
> srtp: use SRTP_SRCTP_INDEX_LEN define from libsrtp 2.6.0
>
> BUG=webrtc:42222036
>
> Change-Id: Ibf5c6b200501c114b9709b76685bb0ecd30bf9fb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359627
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43538}
Bug: webrtc:42222036
Change-Id: Icdac768bd4ccb6f1f4ada68637c0b979aefc39f6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371240
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43544}
removing the webrtc need for having sources in it.
BUG=webrtc:42226155
Change-Id: I40fbde9064f4fa629c7c6b0cf99f23ab1726da75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43540}
srtpfilter was a SDES thing which is gone.
BUG=None
Change-Id: I060582b5ba9e72d1fdad3662e2b478042f0c780c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370640
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43517}
Set use_default_launcher=false in rtc_test on android
Bug: webrtc:42223878
Change-Id: If05da40b420d5da8f9e0f39560eb07380ebada14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368921
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43505}
and misc cleanup
BUG=webrtc:367395350
No-Iwyu: remaining IWYU failure is deep inside gtest which is unrelated to the changes and needs to be investigated separately
Change-Id: I5c2b7a6cc6b15fc5474c55eb98635cb9145b7373
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370180
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43498}
This replaces the WaitUntilCondition function that was used in the
peer_connection_encodings_integrationtest previously. Along with that it
adds tests and improved error message printing.
As a drive-by, matchers were added for RTCError as these are the return
type of this utility function.
Bug: webrtc:381524905
Change-Id: If7ff18692396d3996b5b289f2d2c92520226003e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43494}
This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3.
Reason for revert: Broke internal tests.
Original change's description:
> Use PayloadTypePicker for video PT assignment
>
> This includes changes that change the order of codecs.
> It is preparatory to doing late assignment of video PTs.
>
> Bug: webrtc:360058654
> Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43489}
Bug: webrtc:360058654
Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43490}
This includes changes that change the order of codecs.
It is preparatory to doing late assignment of video PTs.
Bug: webrtc:360058654
Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43489}
also fix a few TODOs
Bug: None
Change-Id: I2d287ed1a58f71ef32d5dc5624879ae8c63348b5
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370122
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43485}
For an offer in sendrecv direction, if for example it can send H.265
level 5.2 while receiving 6.0, setCodecPreferences on offerer's transceiver will currently remove H.265 from the offer SDP, since currently we do a precise level match on send_recv_codecs with the codecs from setCodecPreferences.
Update the matching logic to ignore the level when matching.
Bug: chromium:41480904
Change-Id: Id0f89cbf117ce62249a99257dcce18b35f407cb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369960
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43477}
Cases above 100 ms have been observed on mac; use 60 seconds as
an offset.
Bug: webrtc:380712819
Change-Id: I52a085cb196472188bb5493276a1b32524717c1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369881
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43473}
Two fixes to deflake,
1. Increase the ramp up time for all layers - short time was flaky for
720p.
2. Wait for both the scalability mode AND implementation name to update.
Sometimes the implementation name would change before the scalability
mode did due to a race, so some OutboundRtpStats would have the wrong
values.
To achieve #2 (and #1 with some debugging) a new utility
WaitForCondition was added in order to apply matchers to a condition.
This is used instead of EXPECT_WAIT_EQ and similar because it gives
clear feedback on failure.
I have made 500 runs without a further flake.
Bug: webrtc:381216372
Change-Id: I0132377774e379857664e9a0c20f432bc9dc9fb7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369742
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43472}
Previous computation assumed that local clock is UTC. It isn't.
Adding integration test for abs-capture stats.
Bug: webrtc:380712819
Change-Id: I054d61984cbd017b7ad04ab13e5a687eab89db69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369421
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43465}
These methods previously had a default implementation that triggered
a crash. All implementations must now return a valid object, which
simplifies the code that calls them.
Bug: webrtc:13931
Change-Id: I877fbc929b58c6b83767c6ac5a81c8aa942e3fef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369021
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43453}
The RTCP mode is a send property for both send and receive channels. Send properties should be configured based on what peers support/prefer, which is described by the remote description (content).
Bug: webrtc:340041654
Change-Id: I18cd59e98aecfbbd8f4919b98381836184c10d77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368980
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#43449}
This stores usage for all cases, making it easier to discover
abusive usages on unexpected patterns.
Bug: None
Change-Id: I62c9b07498e811ac04c221f57cfbc02312aaaacc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368902
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43442}
This is preparatory to ensuring that transport-cc gets turned off when
RFC8888 ccfb is negotiated.
Bug: webrtc:378698658
Change-Id: Ie76677bd6aa046701562bbd93d8489858488f863
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368543
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43426}
On a sendrecv m-line, the offered level-id represents the maximum that
can be both sent and received; on a sendonly m-line, the offered
level-id represents the maximum that can be sent; on a recvonly m-line,
the offered level-id represents the maximum that can be received.
Also according to RFC 7798 section 5, the highest level indicated by the
answer is either equal to or lower than that in the offer
Bug: chromium:41480904
Change-Id: I1729c8edc3aed0c00c41cea96204abafc37c002b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367322
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43425}
The gn target for rtc_pc_unittests cleared the "configs" that is by
default set for rtc_test. Restore it back so we get RTC_ENABLE_H265
macro when rtc_use_h265 is configured.
BUG: chromium:41480904
Change-Id: If172482776e5be2ad99d976db12dcfa556ee8a24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368183
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43403}
This will turn on RFC 8888 feedback messages if "ack ccfb" is negotiated.
This should eliminate the need for the "force" flag in the field trial.
Bug: webrtc:42225697
Change-Id: Iec7a894c244a417a8499200861550a33f89966a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367400
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43398}
which launched a while back.
BUG=webrtc:40644399,webrtc:364825888
Change-Id: Ied1d76d8ab2cbb395e09c08f6354d99b4e082cef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367840
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43383}
Inactive encoders are included in the string when they are paused due to
bitrate allocation being 0 for that simulcast layer.
#rtc_ktlo
Bug: webrtc:376804631
Change-Id: I4234b452b60fee58981907380df41962fda5bf40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367660
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43367}
Note: This still doesn't enable CCFB - it just completes the signalling.
Bug: webrtc:42225697
Change-Id: I2dfd346075f2adcc438588f592c8f735f4101c89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367260
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43348}
- Avoid redundant get() when dereferencing smartpointers
- Use const ref instead of copy for RtpExtension
- Use `.empty()` instead of `.size() == 0`
- Remove some unused using declarations
Bug: None
Change-Id: I0dfdc0dfdf165f153c9ba119c115cd492e9599fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367100
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43334}
These functions seem to have been unused except for tests.
It seems to have been added in 2017.
Bug: None
Change-Id: I01983f4b72456b1df27ec2d346014e0de1b5cae7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366943
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43332}
To stress there is no intention to use each instance more than once.
Bug: webrtc:369904700
Change-Id: Id53ad804f39f8ee596ec0b45ff15393009fdfab0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366640
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43324}
It works in the same way as the first packet received callback and can be used for latency measurements.
One important detail is that RTCP and probe packets are excluded from triggering the callback.
Bug: b/375148360
Change-Id: I5f99b565f96b622e864669cf227be5534aab0fc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366644
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43309}
This is a reland of commit 82617ac51e7825db53451818f4d1ad52b69761fd
The reason for the revert was a downstream use of
`rtc::VideoSinkWants::requested_resolution`, so in this reland we don't
rename this field, it's fine just to rename the one in
RtpEncodingParameters for now.
Original change's description:
> Rename `requested_resolution` to `scale_resolution_down_to`.
>
> This is a pure refactor/rename CL without any changes in behavior.
>
> This field is called scaleResolutionDownTo in the spec and JavaScript.
> Let's make C++ match to avoid confusion.
>
> In order not to break downstream during the transition a variable with
> the old name being a pure reference to the renamed attribute is added.
> This means we have to add custom constructors, but we can change this
> back to "= default" when the transition is completed, which should only
> be a couple of CLs away.
>
> Bug: webrtc:375048799
> Change-Id: If755102ccd79d46020ce5b33acd3dfcc29799e47
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366560
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43300}
NOTRY=True
Bug: webrtc:375048799
Change-Id: Ic4ee156c1d50aa36070a8d84059870791dcbbe5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366660
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43304}
