We knew that we should not update buffer level during DTX period. We already fulfill this upon no packet receipt. But we missed doing it for DTX-signaling packets. This CL is to fix that.
Bug: b/129521878
Change-Id: I72ca18e3b21e956123fe6e3119ef0d7c981c9eec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133183
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27643}
Instead use WebRtcKeyValueConfig and FieldTrialBasedConfig
BUG=webrtc:10335
Change-Id: Ie148cb466f86d8fa1ded5c7f125fbcccf6e7dbe3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132714
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27642}
The new processing applies only in Unified Plan mode.
Plan B retains the old-style processing.
This is a reland of 1fa06041bcd8a0119e557d16e7b54a9110c5ad03
Original change's description:
> Make negotiationneeded processing in PeerConnection spec compliant.
>
> This CL fixes the problem of misfired negotiationneeded notifications due
> to the lack of a NegotiationNeeded slot and the proper procedure to
> update it.
>
>
> Change-Id: Ie273c691f11316c9846606446f6cf838226b5d5c
> Bug: chromium:740501
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131283
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27594}
Bug: chromium:740501
Change-Id: I048ae81b2b00086f6d669e94eecf426f0db0ec08
TBR: steveanton@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133162
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27640}
Current logic requires a stream id and track ids when creating a stream
that does not have SSRCs signaled.
This change removes the requirement for stream ids. The requirement for
track id is softer, as one should be generated when it is not present.
Bug: webrtc:10551
Change-Id: Ibc0cc181c6b18efa8394b6c0e4820e3a13da70c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133080
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27633}
As non-linear mode uses a suppressed version of y (not e) as output, this change
uses Y2, rather than E2, as nearend spectrum when computing the suppression gains.
E2 is still used in linear mode.
This change also affects how the minimum suppression gains are calculated. The
minimum gain is now min_echo_power / weighted_residual_echo.
Bug: webrtc:10550
Change-Id: I2904c5a09dd64b06bf25eb5a37c18dab50297794
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133023
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27629}
This CL adds a field trial that enables the EncoderBitrateAdjuster to
allow higher target bitrate if we are not network constrained. We still
don't allow the bitrate to go higher than the average target media rate
though.
Bug: webrtc:10155
Change-Id: Id5995070aa0cbe84b9305a422279141b38664bb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132717
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27627}
Instead use WebRtcKeyValueConfig and FieldTrialBasedConfig.
The purpose is to allow a user of GoogCC to use different settings on different instances.
BUG=webrtc:10335
Change-Id: I2f837688c9fdd341eecb44484cc784b1c80da1a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132791
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27617}
This allows picking up the output in Android tests, where stdout/stderr
is lost but RTC_LOGs are picked up by the org.webrtc.Logging utility.
Tested: Downstream Android tests.
Bug: webrtc:10349
Change-Id: I1379f4303640dbc9621c64d9c88cf61bc8447ab6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132704
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27616}
This CL adds an experiment where aggressiveness of the rate controller
is tuned based on if the application is network constrained or not.
Bug: webrtc:10155
Change-Id: I6c8cd116f57321c5b36cf5a69840913936091aaa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132786
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27615}
Wav file capturer won't repeat file or produce silence after file end and
WebRTC pipeline will crash in such case. In future we need to make it
possible to continue audio after file was ended to behalf in the same
way as video files capturer.
Bug: webrtc:10138
Change-Id: I35f5bd33790cd430a56002a44af0abb894a96d29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132795
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27609}
The former became redundant and didn't guarantee
numerical stability for variance computation.
Bug: webrtc:10412
Change-Id: Idc291abe9add41bde9e7734f179e5d6c65f2630b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132460
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27605}
We no longer have a need for a HKDF implementation in WebRTC. To keep
code quality high it makes sense to delete this dead code path.
Bug: webrtc:9600
Change-Id: Ibe6ee9150acd9dbf59452372242d857c5ffa65c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132802
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27604}
This patch introduces a minor tweak to how often
the extra ice pings are sent.
- never send if non of the candidates is relay
- only send (extra) if it was more than 100ms
since you sent a ping.
The motivation for this is that we measured
an regression of 0.05% in call setup success rate.
Bug: webrtc:10273
Change-Id: Icff36297d57030853a9ff8d4f74aaf6c84051d26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132702
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27601}
This reverts commit 1fa06041bcd8a0119e557d16e7b54a9110c5ad03.
Reason for revert: Likely cause for breaking downstream projects
Original change's description:
> Make negotiationneeded processing in PeerConnection spec compliant.
>
> This CL fixes the problem of misfired negotiationneeded notifications due
> to the lack of a NegotiationNeeded slot and the proper procedure to
> update it.
>
>
> Change-Id: Ie273c691f11316c9846606446f6cf838226b5d5c
> Bug: chromium:740501
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131283
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27594}
TBR=steveanton@webrtc.org,magjed@webrtc.org,sakal@webrtc.org,hbos@webrtc.org,guidou@webrtc.org
Change-Id: Iad7b7d4e37227fa6a76ff830160ca3da9dbe4719
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:740501
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132761
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27599}
This CL allows audioproc_f to overrule any runtime settings for the
pre-amplifier gain that are present in the aecdump file.
Bug: webrtc:10546
Change-Id: I74dbf8d043f59b516bf0abc80f266e965af0754d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132558
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27598}
