This CL adds [[deprecated]] to the old signatures, and uses the new
signatures throughout.
Bug: webrtc:14870
Change-Id: Ic9a8198ac0a2f954e1b2e7d05a55dbe04342f958
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314962
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40517}
This is part of the long term plan to stop using pointer + length
to pass around buffers.
Bug: webrtc:14870
Change-Id: Ibaf5258fd326b56132b9b5a8a6b1563a763ef2f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314960
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40512}
In the past, only encodings.size() == 1 was considered singlecast. But
it's possible to have singlecast via {active,inactive,inactive} too so
this condition should be updated.
This CL ignores x-goog-max-bitrate if maxBitrate was specified on *any*
encoding. This fixes the case of {active,inactive,inactive} resolving
the singlecast inconsistency, but it also takes things one step further
and ignores x-goog-max-bitrate in simulcast cases as well (if any
active encoding has a maxBitrate), as it is not clear why simulcast
should behave differently from singlecast with regards to this flag.
Bug: webrtc:15390
Change-Id: If89a488249239a6bd10fdd56c599ccd2e6ec26fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313540
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40494}
EncoderStreamFactory has two code paths for creating a stream: the
"simulcast path" and the "default path". Only the former cares about
encoding paramter's maxBitrate. The latter assumes that
`encoder_config.max_bitrate_bps` already encompasses the maxBitrate of
the first encoding, but this is not always the case.
As of M113, when scalability mode is specified, {active,inactive} does
not count as simulcast stream but as a default stream represented by
encoding[0].
The problem is that `encoder_config.max_bitrate_bps` only includes
`encodings[0].max_bitrate_bps` when `encodings.size() == 1` which isn't
the case here.
This CL fixes the problem by making the "create default stream" code
path look at the first encoding's maxBitrate and remove existing
assumptions that `encoder_config.max_bitrate_bps` encompasses
`encodings[0].max_bitrate_bps`. This is a step in the right direction
since we're trying to remove all special cases and have encodings map
1:1 with SSRCs, so the "max bps of entire stream" should indeed be a
separate limit than the per-encoding limits and it was confusing that
sometimes it included and sometimes it excluded encoding[0]'s limit.
This issue did not happen in {inactive,active} since that code path
counts as "simulcast stream", so "default stream" is only ever
applicable for index 0.
TESTED=Simulcast Playground, see https://crbug.com/1455962.
Bug: chromium:1455962
Change-Id: I7c44925b780623b5979751e8959e972293648a3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313282
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40482}
ReconfigureEncoder() is supposed to recreate the send stream when
switching between legacy and standard API paths to ensure that the
upper and lower layers agree on the number of streams that exist
(legacy = 3 encodings but 1 stream, standard = same as encodings).
This successfully happened when going from standard to legacy but due
to a bug in the condition this did not happen when going from legacy to
standard because `scalability_mode_used` is always false here (even
though the standard path does use a scalability mode).
As a consequence, SetRtpParameters()'s call to UpdateSendState()
resulted in a DCHECK-crash. In release builds we still avoid IOOB
because active_modules.size() < rtp_streams.size() but to avoid mistakes
like this happening again in the future, the DCHECK is promoted to a
CHECK.
The fix is to remove the scalability mode condition which didn't make
sense anyway - changing scalability mode does not require recreation but
recreation is necessary when number of streams change, whether or not
scalability mode changed.
TESTED = Using Simulcast Playground and switching back and forth
between standard and legacy and changing scalability modes and
confirming from stats, see https://crbug.com/1467455.
Bug: chromium:1467455
Change-Id: Ide29742972ba83f2e0a11f135ab9b39c39d4eb49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313280
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40477}
This reverts commit ebf71114a326080d523b3bc0c2160b2d848d8910.
Reason for revert: Breaks downstream project.
Original change's description:
> Clean up WebRTC-FilterAbsSendTimeExtension field trial
>
> which has been enabled by default for a while. Also document the
> expected behavior, see
> https://groups.google.com/g/discuss-webrtc/c/vfrnxWBVcdA/m/ASf7dBJOGAAJ
> for more details.
>
> BUG=webrtc:10234
>
> Change-Id: If793e2b4b6cebb07371bfdf1f94ed8d49bf2bb34
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311281
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Konrad Hofbauer <hofbauer@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40417}
BUG=webrtc:10234
Change-Id: I856991260ff40a24f03f6054a5c2a9e6f37f47da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311803
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40438}
which has been enabled by default for a while. Also document the
expected behavior, see
https://groups.google.com/g/discuss-webrtc/c/vfrnxWBVcdA/m/ASf7dBJOGAAJ
for more details.
BUG=webrtc:10234
Change-Id: If793e2b4b6cebb07371bfdf1f94ed8d49bf2bb34
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311281
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Konrad Hofbauer <hofbauer@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40417}
since this has been shipping receive-only enabled by default since M92.
Sending remains behind a field trial.
BUG=webrtc:8151
Change-Id: Ia44f8b9cf89ee4878074d1469413d847621ce5ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310040
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40377}
This allows to remove some calls to CreateMediaChannel
in the RtpTransceiver code.
This removes the fake engines owning the channels and moves
the responsibility to the tests themselves as it's quite
hard to both return a unique_ptr to a channel and still own it.
The various channel getters from the fake engine are thus
also removed and tests updated accordingly, the channel is
retrieved from internal structs in the tests by going
through the RtpTransceiver objects as it's not possible to
safely get the channels from only a sender or receiver.
As some tests are running in both PlanB and Unified Plan,
getting a transceiver is not working for PlanB. As PlanB
has been deprecated and will eventually be removed,
the problematic tests have either been removed or updated
to only run with Unified Plan.
Bug: webrtc:13931
Change-Id: I0571beca8b9ef2f2089d500802b7b124268d9de3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310340
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40366}
to ensure consistency for both FID and FEC-FR ssrc-groups.
BUG=chromium:1454860
Change-Id: I61277e73e0a28f5773260ec62c268bdc8c2cd738
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309760
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40347}
which means this will not show up in getStats inbound-rtp/outbound-rtp
until the encoder/decoder is known. This has implications in particular
for inbound-rtp where the value is currently "unknown" until video
frames have been received.
This is safe to change as the previous change to gate
decoderImplementation behind getUserMedia access already broke
the assumption that the field is always string.
BUG=webrtc:14906
Change-Id: Ie6040ada3656e80f792c0c32c1b86ad1d6609d3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293600
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40334}
Media*Channel objects used to subclass webrtc::Transport.
This was not an optimal design. This CL makes the transport
a member variable of MediaChannelUtil.
Bug: None
Change-Id: I85d33cc1b32b931e563b7bb2d277f1c512600831
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309800
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40328}
specified in https://github.com/w3c/webrtc-stats/pull/762
and take FlexFEC into account for receive statistics.
BUG=webrtc:15250
Change-Id: Id85775ab1f29487d5b8bf478da6e22071005901a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294881
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40325}
This groups functions for WebRtcVideoSendChannel and
WebRtcVideoReceiveChannel together, rather than interspersing them.
Bug: webrtc:13931
Change-Id: Iecb5bac18e1d370331e9eb546c6b2fde4d92963f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309460
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40312}
Added as stopped by default as it should be requested by the application,
but it should be listed as available.
Bug: webrtc:14631
Change-Id: I301cfd29c79083c97b4a43b8fdafee2dbe4887a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308824
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40300}
This is a reland of commit 49ace8b6548cda6d4ba74abfca9b616f56dbf9bc
Original change's description:
> Merge the codec types
>
> This allows simplifying code in the codebase to be able to remove a lot
> of templated code and special casing for either AudioCodec and VideoCodec.
> Code simplifications will come in later changes.
>
> Bug: webrtc:15214
> Change-Id: I6e75e6ea725163feb6cc4eb49f37b4722d6c6689
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308501
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40276}
Bug: webrtc:15214
Change-Id: I123d1134a212f65cfbc90ecec9013d0aafebd9ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308721
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40294}
This reverts commit 49ace8b6548cda6d4ba74abfca9b616f56dbf9bc.
Reason for revert: Breaks downstream projects
Original change's description:
> Merge the codec types
>
> This allows simplifying code in the codebase to be able to remove a lot
> of templated code and special casing for either AudioCodec and VideoCodec.
> Code simplifications will come in later changes.
>
> Bug: webrtc:15214
> Change-Id: I6e75e6ea725163feb6cc4eb49f37b4722d6c6689
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308501
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40276}
Bug: webrtc:15214
Change-Id: I57778cccc3a13eb9f955f6ece054dee0ff5a7e92
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308720
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40278}
In M113 we made it possible to opt-in to spec-compliant VP9 using
scalabilityMode and scaleResolutionDownBy. Since this would change
behavior in some edge cases a kill-switch flag was also added.
It turns out it was not needed (current Stable: M114) so we can remove
the flag.
Bug: webrtc:14884
Change-Id: Ie3006164c4d6e90acad1d1f4df2fe2b6e3cb2c35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308683
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40277}
This allows simplifying code in the codebase to be able to remove a lot
of templated code and special casing for either AudioCodec and VideoCodec.
Code simplifications will come in later changes.
Bug: webrtc:15214
Change-Id: I6e75e6ea725163feb6cc4eb49f37b4722d6c6689
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308501
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40276}
into separate sections for each implemented class.
Bug: webrtc:13931
Change-Id: I600f49f3fb195761d13d304f112f36c7c62689df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308120
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40260}
Pushing it to the list of extensions to negotiate could result
in enabling it in production.
BUG=None
Change-Id: I98599e9fbac7e2b81b3f2ad0c7759bb052d9d9d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306101
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40250}
It is not used any more.
Bug: webrtc:13931
Change-Id: I266de41abe239907c6d65f4b182a8dc3aacaba3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308022
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40234}
The preferred method to create codecs is to use the function
cricket::CreateAudioCodec or cricketCreateVideoCodec.
Empty codec objects are deprecated and should be replaced
with alternatives such as methods returning an
absl::optional object instead.
Bug: webrtc:15214
Change-Id: I7fe40f64673cd407830dbbb0e541b85a3aee93aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307521
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40226}
Later CLs will switch to these functions, and eventually the
CreateMediaChannel will be deprecated and removed.
Bug: webrtc:13931
Change-Id: I4c5ab89659a47a501728cac217bb1a877fa50047
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307800
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40221}
Part 1 of the migration towards merging the types.
Any method that could belong to the Codec type was moved, the others
are deprecated.
Alternatives to the AudioCodec and VideoCodec constructors are introduced
to allow creating objects of an indefinite type without having to
reference the old classes.
Bug: webrtc:15214
Change-Id: I20e1aa32962821cad98e9a92c2ec86f8f75e5dd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40213}
This ensures that the MediaChannel interface is only implemented
through a send/receive shim, splitting channels also when kBoth is
used.
Bug: webrtc:13931
Change-Id: Ie97809597eaae7b4f504939339795432c34e56cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307461
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40210}
This completes the split-channel work for the Video side.
Note: For ease of review, the implementations in the .cc
file have not been sorted between sender and receiver. This
can be done in a later purely-editorial CL.
Bug: webrtc:13931
Change-Id: I36cf015d5facb1eed368070cb204a8763ac19a9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307180
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40207}
Add an API to pass AudioFrameProcessor as a unique_ptr.
Bug: webrtc:15111
Change-Id: I4cefa35399c05c6e81c496e0b0387b95809bd8f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301984
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40187}
This allows subclasses of MediaSendChannel and MediaReceiveChannel
to derive from MediaChannelUtil without promising to implement
the interfaces.
Bug: webrtc:13931
Change-Id: I998de7566b343032c83cd6e5419f49349f41035f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307140
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40185}
This allows us to decouple implementation classes from the
MediaChannel class.
Bug: webrtc:13931
Change-Id: I22f166cac17c344f943a0382048e8086a193affa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307000
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40179}
The intent is that this object can be used instead of VideoMediaChannel,
clearing the way for decomposing VideoMediaChannel into send and
receive classes.
This CL uses it for the "both" role of WebRtcVideoEngine::CreateMediaChannel; a later CL will use it for all roles on all engines.
Bug: webrtc:13931
Change-Id: Ibd0ca2c3c45b5e3bfcced8f7e30a1edd63cf7654
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306720
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40173}
There are no common functions between MediaSendChannelInterface
and MediaReceiveChannelInterface except media_type().
This allows us to remove the common superclass for the two interfaces,
making for a simpler class structure.
Bug: webrtc:13931
Change-Id: I82a12ca31f0dc62d7bd97bdda34ca37e59a5fd55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306660
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40154}
as the killswitch is no longer required.
BUG=webrtc:12194
Change-Id: Icb825012c50a93ec4dae49be5732d9e4c0adb89d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306182
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40149}
This change allows us to remove one static_cast from tests that
was problematic for another refactoring.
Bug: webrtc:13931
Change-Id: I8e1b5cecadd806b266b6c115b56b18b9613cbe82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306500
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40144}
Prior to this CL, the video `jitterBufferDelay` stat was the accumulated current delay, which is a smoothened version of the target delay. This is not correct according to the spec [1]. Rather, the stat should be the accumulated time spent in the jitter buffer, for all emitted frames. This CL fixes this spec compliance problem.
Expect changes to test metrics and product monitoring as this CL rolls out.
[1]: https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferdelay
Tested:
1. Go to https://jsfiddle.net/jib1/0L6duga2/show
2. Apply 2.0 seconds of video delay.
3. Notice that "Video jitter buffer delay" is slightly less than 1990ms. (2000ms playoutdelayhint - 10ms render delay - Xms decode delay).
Bug: webrtc:15085
Change-Id: I42805faafd7dd3bcdcf3ad08e751e08d6de38906
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304521
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40138}
These tests verify the ability to override either the old or the
new function, and get the expected results.
Bug: webrtc:13931
Change-Id: Iebd0c929eda73dea75f32b96eb91a64e059a3cf8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294880
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40120}
