The tests shows that using a scale factor around 0.5 gives the best precision and F1 score.
Bug: webrtc:358039777
Change-Id: I22557eb9e6318ecaa726b56d3ccb2125fdf65f7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367681
Reviewed-by: Fanny Linderborg <linderborg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Cr-Commit-Position: refs/heads/main@{#43366}
- With mono encoding and stereo decoding check that the decoded
signal is trivial stereo
- DTX tests
- With mono encoding and stereo decoding check that the comfort
noise generated by Opus is NOT(*) trivially stereo
- With stereo encoding and stereo decoding check that the comfort
noise generated by Opus is not trivially stereo
*: the test shows the behavior described in [1] and that needs to
be fixed.
[1] https://issues.webrtc.org/376493209
Bug: webrtc:376493209
Change-Id: I34aacd4bd7c79be9df05c242e912c9981896a73d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367206
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43363}
This is to avoid a dependency on Chromium //base on bots that don't need it:
1bd0da6657:libfuzzer/BUILD.gn;l=164
Bug: None
Change-Id: Idf3ef2a313641abcd3741e0ef7b2fac61c629068
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367640
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43360}
Cleanup some of the TODOs for H.265. They are either invalid or their handling should be merged with other codec types.
Bug: chromium:41480904
Change-Id: I76263354b1b87035e240d77283b21a9a26dcb45b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366044
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43359}
Users can use BuiltinAudioProcessingBuilder directly instead and thus depend on field trials more explicetly.
Bug: webrtc:369904700
Change-Id: I100e73785ebf9fbfcdd80152b6d094a93498d711
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367261
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43356}
The libevent task queue has been migrated to stdlib on Android,
without issues. A month has passed without any feedback. The next
step is to disable usage from the rest of the consumers.
Bug: webrtc:42224654
Change-Id: Iba12cfb1a7c0533c87e4c03f65c5377010b9831e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367480
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43354}
This aligns with current transport sequence number feedback
Bug: webrtc:42225697, b/377028537
Change-Id: I9d3bcc2e131f1a2c20d5f8c3fe5776268b97e00a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367386
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43352}
Fallback to squared pixel generator if no camera exist.
Remove render scaling and rendering of local preview since contains
bugs that crash the demo. Use rtc::Buffer for storing the rendered
frames.
Run build cleaner
Bug: none
Change-Id: I46dc972eaa50069433d8afaa1fe38380edd3d337
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367120
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43350}
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}
These function were replaced with AbslStringify
Bug: None
Change-Id: Ia34b98ed4e0ed18bb52fe9370cff7a6f70caae6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364621
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43346}
- WebRTC does use the libopus DTX implementation
- The removed detail is anyways irrelevant in a docstring
Bug: webrtc:376493209
Change-Id: I3dfe1521259e596dbfa0db97f91ffb75deeb16b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367200
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43344}
With L4S in WebRTC, only RTP packets are supposed to be send with ECT(1)
Bug: webrtc:42225697
Change-Id: If10bf74a867d3ea04fd1fb931cdc2a6380176270
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43343}
New api ensures field trials are available at construction time of the AudioProcessing object.
This would allow AudioProcessing implementation to use propagated field trials during construction.
Also, short term, it ensures AudioProcessing is constructed after global field trials are set.
Bug: webrtc:369904700
Change-Id: If3d00c8a3a509299cd0915d55f13a9a3ce4a7140
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367201
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43340}
and modify DEPS files accordingly.
This is done in support of the decision to encourage AbslStringify.
Bug: None
Change-Id: I26fee77978d1dd21be6d2ef011c4dfd78a7b43e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367204
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43338}
Increased the number of errors the automation is fixing to 150 from
75 in this commit.
Bug: webrtc:370878648
Change-Id: If6e6a5f40db7eb54c27c1a85fb7031838e478c70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366205
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#43337}
- 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}
and remove the testing of the nondefault key size from the "server" parameters to speed up tests
BUG=webrtc:375552698
Change-Id: Ibc1bd491300964aa45826b98962ed3e56c6d4974
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366941
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43321}
It is used to distinguish between audio/video packets and everything else (retransmit/padding/fec), so naming it is_media makes more sense.
This is a follow up to https://webrtc-review.googlesource.com/366644
Bug: b/375148360
Change-Id: Ia53f4d707ceb85f059688d86bc5dcc2d57908d88
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366424
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43319}
After landing this change, we can change the corresponding usage in
blink to start using presentation_timestamp as well and then delete
the remaining usage of capture_time_identifier.
Bug: webrtc:373365537
Change-Id: I0c4f2b6b3822df42d6e3387df2c243c3684d8a41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365640
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#43317}
