This reverts commit b7239a9dc825ddb25dbc800aed3a065163b9a10e.
Reason for revert: Broke chromium mac build, compilation failures on content/renderer/media/webrtc/webrtc_video_frame_adapter.h.
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
TBR=kwiberg@webrtc.org,nisse@webrtc.org
Change-Id: I7334597cc8979ba9cfaff526967084349ef27f3c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8270
Reviewed-on: https://webrtc-review.googlesource.com/5800
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20107}
The refcount.h file doesn't depend on anything from
refcountedobject.h. The motivation of this change to make it possible
to add additional declarations to refcount.h, and include it from
refcountedobject.h.
Bug: webrtc:8270
Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
Reviewed-on: https://webrtc-review.googlesource.com/5760
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20106}
This enables the bit exactness tests for the audio level controller.
Additionally, some expected test values are updated.
Bug: webrtc:8309
Change-Id: Ia73f2a16aea4b3e926d70d8b4b8e5d5d801833c7
Reviewed-on: https://webrtc-review.googlesource.com/4426
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20102}
This CL fine-tunes the internal AEC3 parameters to increase the
transparency of the nearend signal.
Bug: webrtc:8322
Change-Id: I2e35165082d88b8f2b1e8367d8ed0e29bd67b4e5
Reviewed-on: https://webrtc-review.googlesource.com/5365
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20082}
This CL bounds the supppression gain for higher frequencies where
the estimate of the residual echo sometimes is less accurate.
Bug: webrtc:8320
Change-Id: I02b21e6b1758c7e8b6660c1631a05c956a45e4c8
Reviewed-on: https://webrtc-review.googlesource.com/5260
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20081}
Allow a custom version of audioproc_f in APM-QA.
Bug: webrtc:7494
Change-Id: Id9adffd63927202d868bc2fc8b6a54c8e6b07039
Reviewed-on: https://webrtc-review.googlesource.com/4060
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20033}
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}
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 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}
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}
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 https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
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}
* scalar optimization, vectorization (including new file for SSE2 code
and path selection mechanism).
* 0.5% AEC overall speedup for the straight C path.
* 3.0% AEC overall speedup for the SSE2 path.
Review URL: http://webrtc-codereview.appspot.com/46005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@134 4adac7df-926f-26a2-2b94-8c16560cd09d
Lots of AEC CPU usage is coming from calls to 'rdft'. To optimize this,
deep changes (modification of memory layout, ...) have to be done and it
is not practical to do them in an utility library. Most of these changes
will occur in subsequent CLs.
The new file 'aec_core_rdft.c' is a copy of 'modules/audio_processing/
utility/fft4g.c' whose size has been significantly reduced by removing
all code non-necessary to compute rdft. The main entry point and utility
functions have also been modified to take into account the fact that all
'rdft' calls performed by AEC have a length of 128. This yields:
* 1.8% AEC overall speedup for the straight C path.
* 2.3% AEC overall speedup for the SSE2 path.
Review URL: http://webrtc-codereview.appspot.com/44008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@126 4adac7df-926f-26a2-2b94-8c16560cd09d
* new file for SSE2 code, code selection through function pointers.
* structure change for array of complex numbers.
* 3.8% AEC overall speedup for straight C path.
* 8.8% AEC overall speedup for SSE2 path.
Review URL: http://webrtc-codereview.appspot.com/34002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@36 4adac7df-926f-26a2-2b94-8c16560cd09d
* new file for SSE2 code, code selection through function pointers.
* structure change for array of complex numbers.
* 3.8% AEC overall speedup for straight C path.
* 8.8% AEC overall speedup for SSE2 path.
Review URL: http://webrtc-codereview.appspot.com/33003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@26 4adac7df-926f-26a2-2b94-8c16560cd09d
