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}
Making a long chain of interface changes to route a CodecSpecificInfo
struct from the video encoder function to the RTPSenderVideo. This
will be used to convey information needed by the RTP packetizer when
building the RTP headers.
Review URL: http://webrtc-codereview.appspot.com/56001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@140 4adac7df-926f-26a2-2b94-8c16560cd09d
* 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
Added metrics averaged over intervals of the loss/bandwidth reports, to be used for adjustment of robustness settings. Separated this set
from the (global) metrics used for resolution adaptation.
Some code cleanup in content_metrics.cc/.h.
Review URL: http://webrtc-codereview.appspot.com/52002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@125 4adac7df-926f-26a2-2b94-8c16560cd09d
