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}
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}
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 63d03f586b
Original change's description:
> Unify access to SDP codec parameters
>
> which come from the a=fmtp:<pt> lines in the SDP and were used as either
> std::map<std::string, std:string>
> with three aliases,
> cricket::CodecParameterMap
> SdpAudioFormat::Parameters
> SdpVideoFormat::Parameters
>
> Use webrtc::CodecParameterMap in all places.
>
> BUG=None
>
> Change-Id: If47692bde7347834c349c6539b43309d8770e67b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
> 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@{#41375}
Bug: None
Change-Id: I5f8f45688df232eb37b12fa3e56a893a1c754e17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331402
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@{#41467}
which come from the a=fmtp:<pt> lines in the SDP and were used as either
std::map<std::string, std:string>
with three aliases,
cricket::CodecParameterMap
SdpAudioFormat::Parameters
SdpVideoFormat::Parameters
Use webrtc::CodecParameterMap in all places.
BUG=None
Change-Id: If47692bde7347834c349c6539b43309d8770e67b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
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@{#41375}
while cleaning up Call factory function,
- pick rtp_transport_controller_send_factory based on presence in the config instead of based on the call site thus removing one extra factory function.
- when Call is created through test helper TimeControllerBasedFactory use original media factory instead of direct factory, thus allow to configure degraded call through field trials in tests, and ensure difference with production code path stay minimal in the future.
Bug: webrtc:15656
Change-Id: If9c2a9fc871e139502db2bec0a241d8d64c53720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330061
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41329}
Replace CallFactory class with a factory function
Bug: webrtc:15574
Change-Id: Ib1d8cff8d7550da3af01693a7bc117a7bd342258
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330000
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41321}
To be submitted after downstream usage has been removed, but no earlier than December 1, 2023.
Bug: webrtc:12598
Change-Id: Id9acbac591c48c0c5883fe8f06cf6a68471b70f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323004
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41290}
* Pass codec factories to the video codec tester instead of creating and wrapping codecs into a tester-specific wrappers in video_codec_test.cc. The motivation for this change is to simplify the tests by moving complexity to the tester.
* Merge codec stats and analysis into the tester and move the tester. The merge fixes circular deps issues. Modularization is not strictly needed for testing framework like the video codec tester. It is still possible to unit test underlaying modules with rather small overhead.
* Move the video codec tester from api/ to test/. test/ is accessible from outside of WebRTC which enables reusing the tester in downstream projects.
Test output ~matches before and after this refactoring. There is a small difference that is caused by changes in qpMax: 63 -> 56 (kDefaultVideoMaxQpVpx). 56 is what WebRTC uses by default for VPx/AV1 encoders.
Bug: webrtc:14852
Change-Id: I762707b7144fcff870119ad741ebe7091ea109ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327260
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41144}
This is a reland of commit 3ea9fc4cd8
Original change's description:
> Make frame transformer MimeType pure virtual again
>
> after both audio and video have been implemented.
>
> BUG=webrtc:15579
>
> Change-Id: Ib52e8f67292259cbf7497a884672de72f3003282
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326162
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Tony Herre <herre@google.com>
> Cr-Commit-Position: refs/heads/main@{#41114}
BUG=webrtc:15579
Change-Id: Ia020149cba3045022b539f290565d6c1d0e813ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326880
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41121}
after both audio and video have been implemented.
BUG=webrtc:15579
Change-Id: Ib52e8f67292259cbf7497a884672de72f3003282
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326162
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#41114}
To replace CreateTimeControllerBasedCallFactory
Update webrtc tests to use this new function
Bug: webrtc:15574
Change-Id: I2b74cd930ecc4f72dd1e7aa853764ca298b66ad8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325527
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41076}
Remove EncodedFrame::MissingFrame, as it was always false in actual
in-use code anyway, and remove usages of the Decode missing_frames param
within WebRTC. Uses/overrides in other projects will be cleaned up
shortly, allowing that variant to be removed from the interface.
Bug: webrtc:15444
Change-Id: Id299d82e441a351deff81c0f2812707a985d23d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317802
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#40662}
Move the SetRTPTimestamp method from TransformableAudioFrameInterface
to the base class, so that RTPTimestamps can also be modified on encoded
video frames.
Bug: webrtc:14709
Change-Id: I355be527c2be201c9201e04c431394c962237140
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310781
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40378}
Instead switch to specific getters, or methods only defined on specific implementations rather than part of the public API.
Once uses are removed from Chromium, I'll mark GetHeader() deprecated
and eventually remove it.
Bug: chromium:1456628
Change-Id: I19b80489b3a0322c201e24994494cfbb742ee13e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309780
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40344}
Make outgoing encoded audio frames inherit from the same Audio interface
that incoming frames inherit from, to align them and make it possible to
eg clone frames regardless of their direction.
Also begin removing GetHeader() from the Audio interface, replacing it
with getters for the specific values we actually need to propagate in
the API: sequence number and CSRCs. This makes it much easier to treat
incoming and outgoing frames the same, even if they don't have full
RtpHeaders prepared at the point of the transform.
Bug: chromium:1453226
Change-Id: Ib5b39b30dea8a378b3b26efb1589dfd64741d201
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308141
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40309}
This can be done now as the function SetRTPTimestamp is now overriden
in blink MockTransformableAudioFrame.
Change-Id: I4fa4cb81d0282fea864818f0f2d9a5ed881a5d30
Bug: webrtc:14709
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308361
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40257}
This change will make it possible to let us modify timestamp in
RTCEncodedAudioFrame.
Change-Id: I97e9571c258fd718d6c211014f1476ca46c78097
Bug: webrtc:14709
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307501
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40238}
Set of codecs for testing is hardcoded to AV1, VP8, VP9, H264, H265. Some codecs may not be available due to lack of support on the platform or due to some issue in our code which would be a regression. Reporting zero metrics for failed tests would allow the perf tool to detect such a regression.
This also enables codec tests by default. The tests should not run on bots since video_codec_perf_tests binary is not included in any test suits yet.
Bug: webrtc:14852
Change-Id: I967160069055036f93e595d328c4d5f1ca483be9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300868
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39840}
One problem with the existing Send() method is that it has a return
value that is problematic for a fully async implementation.
A second problem with Send() is that the return value is bool and not
RTCError (webrtc:13289), which is why OnSendComplete() uses RTCError.
Also, start deprecating `bool Send()` in favor of `void SendAsync()` and
adding `network_safety_` flag for posting async operations to the
network thread. This flag also takes over from the
`connected_to_transport_` which can now be removed.
Bug: webrtc:11547, webrtc:13289
Change-Id: I87bbc7e9b964a52684bdfe0e6ebc5230be254e8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299760
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39817}
The Mode is currently redundant with the optional input_file_name.
Change-Id: Ib4f0a363e86d925107d61867a7f743d6663e7071
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298743
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39754}
Also make it possible to pause an already paused stream by making it a no-op.
Change-Id: Id10f74a4c6464067ae63208162194f020c6470eb
Bug: b/271542055
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298202
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39620}
Initialization of Android HW codecs takes hundreds milliseconds. Exclude this time from frame processing time of first frame by initializing codecs before starting encoding/decoding.
Bug: b/261160916, webrtc:14852
Change-Id: I9ec84c6b12c1d9821b59965cf521170224066563
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298304
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39613}
following spec updates from
https://github.com/w3c/webrtc-extensions/pull/142
BUG=chromium:1051821
Change-Id: I1fd991a5024d38ac59ebe510ea1a48fd6f42d23b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296321
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39491}
This will clone an encoded audio frame into a sender frame.
Bug: webrtc:14949
Change-Id: Ie62d9f5ec457541b335bde8f2f6e9b6d24704cf6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294560
Commit-Queue: Tove Petersson <tovep@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39480}
It was marked deprecated on Feb 9th, ~3 weeks ago.
Bug: chromium:1414370
Change-Id: I251b91984ca9a958e221f6eaf01c63b05c5a7a48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295506
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39422}
Add a method to TransformableVideoFrameInterface which returns a new
instance of VideoFrameMetadata which the caller can move and use as
they like.
This will replace the existing GetMetadata which returns a dangerous const ref to a field which might change if someone calls SetMetadata
etc. That method will be deprecated as soon as we've migrated Chromium
usages.
Bug: webrtc:14708
Change-Id: Id7c15f33d6ec28c4a975ce250cdc791d7a3087bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Tove Petersson <tovep@google.com>
Cr-Commit-Position: refs/heads/main@{#39403}
Allow mocking of GetDirection in tests.
Required for Chromium adoption of this mock:
https://chromium-review.googlesource.com/c/chromium/src/+/4236916
Bug: chromium:1414370
Change-Id: I2e7443a1bf24966cfcfaeadf47c5b29375e84f99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293745
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39329}
In Meta we have our own audio encoder/decoder factories and we would like to exercise those in the peer connection e2e test framework.
Also, looks cleaner to have the APIs for both video and audio :)
Bug: webrtc:14910
Change-Id: Ibd1e0f39fc809882ef17b3de3154fdf4b567013b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293782
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39328}
Also move the frame_transformer_factory_unittest build target into the
if(rtc_include_tests) block, so it's not compiled without the mock.
Bug: chromium:1414370
Change-Id: I12653b173b419ec20bfad904e24a4d965e7e7830
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292863
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39288}
Passing of ownership of codecs to tester is not strictly needed. We may need to continue using a codec after test. For example, to check codec state or to use the same codec instance in next test.
Bug: b/261160916, webrtc:14852
Change-Id: I179b262116d7de76b8171f0409f943ad6d87433e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291802
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39256}
This CL introduces VideoCodecStats and VideoCodecStatsImpl which provide baseline functionalities for storing, slicing and aggregation of encoded and/or decoded video frame statistics. To facilitate metrics logging (not implemented yet), SamplesStatsCounter is used for stream parameters.
VideoCodecStats/VideoCodecStatsImpl will replace existing VideoCodecTestStats/VideoCodecTestStatsImpl.
Bug: b/261160916, webrtc:14852
Change-Id: I0f96ce1ed9be3aee2a702804612524676c9882fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291323
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39248}
RTPVideoHeader is changed to non-const to allow modifying it. We want
to do this when implementing setMetadata() in JavaScript or when
refactoring clone() as "construct + set bytes + setMetadata".
Unblocks
https://chromium-review.googlesource.com/c/chromium/src/+/4164979.
Bug: webrtc:14709
Change-Id: I6089df9c03e9aa33feeb0830dd240dd456cb565e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290981
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39113}
Keeping the headers to allow compatibility with current users
that expect the headers to be in that target before they are
also updated.
Bug: webrtc:9838
Change-Id: I8b1e88850958e92c043686587a37791f01860220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39031}
This tester is an improved version of VideoProcessor and VideoCodecTestFixture and will eventually replace them.
The tester provides better separation between codecs and testing logic. Its knowledge about codecs is limited to frame encode/decode calls and frame ready callbacks. Instantiation and configuration of codecs are the test responsibilities.
Other differences:
- Run encoding and decoding in separate threads
- Run quality analysis in a separate thread
- Reference frame buffering is moved into video source (which re-read frames from the file).
- Make it possible to run decode-only tests
This CL is MVP implementation: it adds only 1 test (video_codec_test.cc, ConstantRate/EncodeDecodeTest) and the test is disabled for now.
Bug: b/261160916
Change-Id: Ida24a2fca1b1496237fa695c812084877c76379f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283525
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38901}
Passes frame_types to the underlying encoder in bypass mode.
For libvpx this has no effect, for H264 this changes the behavior
to allow generating keyframes on a per-layer basis.
BUG=chromium:1354101
Change-Id: I26fc22d9e2ec4681a57ce591e9eafd0b1ec962b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285083
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38821}
Extends the mocks for rtpreceiver rtpsender and videotrack. This change
allows the external HangoutsKit client to remove its own mocks of rtc
types.
Bug: none
Change-Id: I8ba1752fe7633f9e0bba264a1279f74cc1368a2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282900
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jack Smith <jackdsmith@google.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38782}
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.
Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
