The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic.
The simulation is done by applying digital gain to the input samples.
This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal.
This CL has been ported from https://codereview.webrtc.org/2834643002/.
Bug: webrtc:7494
Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48
Reviewed-on: https://webrtc-review.googlesource.com/2685
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19992}
We are doing some unconventional stuff in jni_generator_helper.h in
order to integrate the Chromium script with WebRTC. Long term, we will
improve this and remove the lint suppressions.
Bug: webrtc:8278
Change-Id: I5d6f0017c4deab4586844647f7cd657641fecbab
Reviewed-on: https://webrtc-review.googlesource.com/3780
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19989}
Test often fails on line ortcfactory_integrationtest.cc:321 on bot
iOS64 Debug.
TBR=deadbeef@webrtc.org
NOTRY=True
Bug: webrtc:7915
Change-Id: I4bf8caa13df3fcb416f380f9a64593d00843f3d6
Reviewed-on: https://webrtc-review.googlesource.com/3961
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19988}
A value for this flag was derived in RtpReceiverImpl::IncomingRtpPacket.
For audio, it was never used, and for video, it was overridden by
the result from RtpDepacketizer::ParsedPayload.
Bug: webrtc:7135
Change-Id: I597a57ca77d13b9a9145a9ee5e6624c1986777b9
Reviewed-on: https://webrtc-review.googlesource.com/3660
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19985}
The SrtpTransport takes the SRTP responsibilities from the BaseChannel
and SrtpFilter. SrtpTransport is now responsible for setting the crypto
keys, protecting and unprotecting the packets. SrtpTransport doesn't
know if the keys are from SDES or DTLS handshake.
BaseChannel is now only responsible setting the offer/answer for SDES
or extracting the key from DtlsTransport and configuring the
SrtpTransport.
SrtpFilter is used by BaseChannel as a helper for SDES negotiation.
BUG=webrtc:7013
Change-Id: If61489dfbdf23481a1f1831ad181fbf45eaadb3e
Reviewed-on: https://webrtc-review.googlesource.com/2560
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19977}
This header will be included from generated JNI code, and acts as a
bridge between JNI types in WebRTC and Chromium.
Bug: webrtc:8278
Change-Id: I88331d26315aa8b258aaaaa26d82324660d648b5
NOPRESUBMIT: True
Reviewed-on: https://webrtc-review.googlesource.com/3441
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19974}
This CL adds an interface for a generic PostProcessing module that
is optionally added to the APM at construction time.
(Parenthetically this CL also adds a missing lock check to
InitializeGainController2.)
Bug: webrtc:8201
Change-Id: I7de64cf8d5335ecec450da8a961660906141d42a
Reviewed-on: https://webrtc-review.googlesource.com/1570
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19973}
In order to compute a THD score, a pure tone must be used as input signal.
Also, its frequency must be known. For this reason, this CL adds a number of
changes in the APM-QA pipeline. More in detail, input signal metadata is loaded
and passed to the THD evaluation score instance. This makes the eval_scores
module less reusable, but it is fine since the module has been specifically
designed for the APM-QA module.
BUG=webrtc:7494
Review-Url: https://codereview.webrtc.org/3010413002
Cr-Commit-Position: refs/heads/master@{#19970}
This is a workaround until downstream projects have been fixed.
BUG=webrtc:8220
Review-Url: https://codereview.webrtc.org/3017613002
Cr-Commit-Position: refs/heads/master@{#19966}
Replaced with scalars ulpfec_payload_type and red_payload_type.
In particular, ulpfec.red_rtx_payload_type, which duplicated info in
rtx_associated_payload_types, is deleted. This is a followup to cl
https://codereview.webrtc.org/3012963002.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/3019453002
Cr-Commit-Position: refs/heads/master@{#19965}
This reverts commit 7a2bfd22e69f14e2af989b9e30ddd834f585caa9.
Reason for revert: Breaks external test.
Original change's description:
> Improve unit testing for HardwareVideoEncoder and fix bugs.
>
> Improves the unit testing for HardwareVideoEncoder and fixes bugs in it.
> The main added feature is support for dynamically switching between
> texture and byte buffer modes.
>
> Bug: webrtc:7760
> Change-Id: Iaffe6b7700047c7d0f9a7b89a6118f6ff932cd9b
> Reviewed-on: https://webrtc-review.googlesource.com/2682
> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#19963}
TBR=magjed@webrtc.org,sakal@webrtc.org
Change-Id: If1e283a8429c994ad061c7a8320d76633bd0d66b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7760
Reviewed-on: https://webrtc-review.googlesource.com/3640
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19964}
Improves the unit testing for HardwareVideoEncoder and fixes bugs in it.
The main added feature is support for dynamically switching between
texture and byte buffer modes.
Bug: webrtc:7760
Change-Id: Iaffe6b7700047c7d0f9a7b89a6118f6ff932cd9b
Reviewed-on: https://webrtc-review.googlesource.com/2682
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19963}
This is a trivial change to add more logs in DX capturer components for
debugging purpose.
Bug: chromium:764258
Change-Id: I406127d838a522f0226720434e840c7163b4719d
Reviewed-on: https://webrtc-review.googlesource.com/3541
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19960}
Adds a new method PopulateDelayManagerStats which takes care of the
fields that needed information from the DelayManager.
Also adds a new test for StatisticsCalculator made practically
feasible by the refactoring.
Bug: webrtc:7554
Change-Id: Iff5cb5e209c276bd2784f2ccf73be8f619b1d955
Reviewed-on: https://webrtc-review.googlesource.com/3181
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19957}
Test was Android-only, so it was disabled completely.
TBR=brandtr@webrtc.org
Bug: webrtc:8280
Change-Id: Id45eedac90fb892f5a380e5c2614037e01ee8c76
Reviewed-on: https://webrtc-review.googlesource.com/3460
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19954}
This annotation will be used to annotate Java classes that are
referenced from native code.
Bug: webrtc:8278
Change-Id: Icf020927d377ba04304ddbf92639e6ef174de22c
Reviewed-on: https://webrtc-review.googlesource.com/3300
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19951}
Moved from https://codereview.webrtc.org/3009093002/
TBR=hlundin-webrtc
Bug: webrtc:8041
Change-Id: I33485629a6f1dcb86fd4242468841605e7d8a72a
Reviewed-on: https://webrtc-review.googlesource.com/3440
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19949}
See https://codereview.webrtc.org/3012183002/ for more background.
Bug: webrtc:8206
Change-Id: I638bc30a44d036826b7caccaab254916093fe357
Reviewed-on: https://webrtc-review.googlesource.com/1584
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19946}
Reason for revert:
Fixes has landed.
Original issue's description:
> Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
>
> Reason for revert:
> We are not certain this is the behavior we want.
>
> Original issue's description:
> > Fix the video buffer size should take rtt into consideration
> >
> > BUG=webrtc:8010
> >
> > Review-Url: https://codereview.webrtc.org/2980413002
> > Cr-Commit-Position: refs/heads/master@{#19285}
> > Committed: f1e08d0b58
>
> TBR=sprang@webrtc.org,gustavogb@gmail.com
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:8010
>
> Review-Url: https://codereview.webrtc.org/3002033002
> Cr-Commit-Position: refs/heads/master@{#19442}
> Committed: bdbc8895f3TBR=sprang@webrtc.org,gustavogb@gmail.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8010
Review-Url: https://codereview.webrtc.org/3016633002
Cr-Commit-Position: refs/heads/master@{#19944}
