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}
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}
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 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}
When NetEq performs a merge operation, it will usually have to correct
the stats for number of concealment samples produced, sometimes with
decreasing it.
This does not make sense in the context of the stats spec, and
stats-consuming applications may not be prepared for it. With this
change, only positive corrections are allowed for the
concealed_samples value. This will sometimes lead to a small positive
bias, but it will be negligible over time.
Bug: webrtc:8253
Change-Id: Ie9de311ab16401f1a4b435f6269725901b8cf561
Reviewed-on: https://webrtc-review.googlesource.com/1583
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19941}
This reverts commit 262d4ff882d62985426d4c31bae1411c7d5ed0e1.
Reason for revert: The logging in this CL is spamming the logs. Therefore I'll revert and reland this once that has been fixed.
Original change's description:
> Added logging inside AEC3 for render API buffer under/overruns
>
> Bug: webrtc:8250
> Change-Id: Ib9ce26419b8961a33869d2f24cc4248fe10039b8
> Reviewed-on: https://webrtc-review.googlesource.com/1562
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#19856}
TBR=gustaf@webrtc.org,peah@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8250
Change-Id: Icbbb219772ca2e3644b9fcb7fa99545b147fd675
Reviewed-on: https://webrtc-review.googlesource.com/2720
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Noah Richards <noahric@chromium.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19932}
- Group member variables into two structs: target rates/actual rates.
- Split verify and print of rate control metrics into separate functions.
- Rename member variables.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/3009423002
Cr-Commit-Position: refs/heads/master@{#19925}
* Add audio_ prefix to CallTest::{en,de}coder_factory_.
* Let VideoQualityTest only instantiate encoders using encoder factories.
* Add HW encoder factories to VideoQualityTest.
* Add full stack tests:
- sqcif7 at 30 kbps: MediaCodec and libvpx.
- 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx.
BUG=webrtc:8219
Review-Url: https://codereview.webrtc.org/3005253002
Cr-Commit-Position: refs/heads/master@{#19923}
It's in the way of a refactoring.
Also change PayloadTypeToPayload---the method all callers can use instead---to return Optional<Payload> instead of const Payload* (for thread safety reasons: an object that protects itself with a mutex shouldn't be handing out pointers to parts of itself).
BUG=webrtc:8159
Change-Id: I7ef0d545077ffdea016b309f2165e3c4955a2928
Reviewed-on: https://webrtc-review.googlesource.com/2360
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19917}
There was one .h file that didn't have to be public. :-)
BUG=webrtc:8159, webrtc:8255
Change-Id: I0998f0340384c57f52affdde30f6b4eb2eaa712b
Reviewed-on: https://webrtc-review.googlesource.com/2400
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19915}
The goal of this CL is to separate Obj-C/Obj-C++ code from targets
which have also C++ code (see
https://bugs.chromium.org/p/webrtc/issues/detail?id=7743 for more
information).
To achieve this we have created 2 targets (desktop_capture_objc and
desktop_capture_generic) and desktop_capture will act as a proxy
between these targets (this way we can avoid a circular dependency
between desktop_capture_generic and desktop_capture_objc).
NOTRY=True
Bug: webrtc:7743
Change-Id: I19f8bb8719cfc6af259819e2089cebea72b5d531
Reviewed-on: https://webrtc-review.googlesource.com/2220
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19899}
After the migration from serc/webrtc to src/ this entry in the
include_dirs list is not needed anymore.
Bug: chromium:611808
Change-Id: I17c87509b73b8a44f758d59ada28d366da664649
Reviewed-on: https://webrtc-review.googlesource.com/1920
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19894}
The number of concealment events. This counter increases every time a concealed sample is
synthesized after a non-concealed sample. That is, multiple consecutive concealed samples
will increase the concealedSamples count multiple times but is a single concealment event.
Bug: webrtc:8246
Change-Id: I7ef404edab765218b1f11e3128072c5391e83049
Reviewed-on: https://webrtc-review.googlesource.com/1221
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19881}
This CL adds an offset to the delay estimation used in AEC3 for
determining the alignment between the render and capture signals.
This ensures that there is no possibility for the capture loss to
cause the delay estimation to miss aligning the signals.
BUG=webrtc:8247, chromium:765242
Change-Id: I526dc7971b13425a28e99d69168fd3722a4cfdae
Reviewed-on: https://webrtc-review.googlesource.com/1232
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19871}
Wrapper pattern is widely used in DesktopCapturer implementations. So this
change adds DesktopCapturerWrapper and CaptureResultDesktopCapturerWrapper as
the base classes of other wrappers. Implementing a new wrapper should become
easy, the implementation does not need to care about the uninteresting
overrides.
Bug: chromium:764258
Change-Id: If91c1b5e778805906f7f77854ea5600aa61bf64a
Reviewed-on: https://webrtc-review.googlesource.com/1420
Commit-Queue: Zijie He <zijiehe@google.com>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19868}
Set limits of NetEq minimum and maximum delay to 0-10000 ms closed interval.
Fixed error message in Audio Coding Module.
Bug: webrtc:6861
Change-Id: Id1b9928f808bdb6e1088c6895f2ec4a53b00efb2
Reviewed-on: https://webrtc-review.googlesource.com/1343
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19860}
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.
The cpplint complaint is:
Include the directory when naming .h files [build/include] [4]
This CL disables the error but we have to remove these two headers
from the root directory.
NOPRESUBMIT=true
Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
