The implementation of this proposal is in progress.
More unittest will be added.
Sender side is being implemented.
Some constants need to be tuned.
BUG=4550
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43299004
Cr-Commit-Position: refs/heads/master@{#9146}
Changes:
1. Documented return values of VoENetwork methods.
2. In VoENetworkImpl: replaced calls to SetLastError() with LOG_F(). SetLastError() is not used anymore because no one is calling LastError() to check for last error. Also, its usage is being removed in Video Engine and we want to be consistent.
3. In VoENetworkImpl: removed WEBRTC_TRACE() usage.
4. In VoENetworkImpl: replaced some defensive code with assert(). We are now assuming that the caller has called VoEBase::Init() where needed. We are also assuming that it is invalid to pass nullptr where data is expected.
5. Updated unit tests accordingly.
R=henrika@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53369004
Cr-Commit-Position: refs/heads/master@{#9145}
When running AEC in extended_filter mode there is no startup phase to evaluate the reported system delay values.
Instead we simply use the first value and scale by two to avoid over compensating when synchronizing render and capture.
We don't need to be too accurate since we have extended the filter length.
On Android we use fixed (measured) reported delay values.
There is no need to be extra conservative here, because that is already built-in in the measured value.
In fact, the difference between devices is large and with such an extra conservative approach the true delay can not be caught by the filter length.
With this change we can improve performance on some devices.
BUG=4472
TESTED=offline on recordings from various devices
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49909004
Cr-Commit-Position: refs/heads/master@{#9144}
The delay estimator has a robust_validation mode used to deliver more stable delay etimates. The cost is increased reaction time when we have a delay jump.
This mode can be turned on and off on the fly, but statistics are not updated while disabled. This makes the estimator unreliable if it is enabled on the fly.
This CL makes sure the update is always done.
BUG=4472
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50889004
Cr-Commit-Position: refs/heads/master@{#9143}
For quick and easy aecdump verifiation storing data as text speeds up the issue tracking process, since anyone can simply view values like mic volume.
BUG=4609
TESTED=verified unpacking an aecdump with flag --txt stores that data in text files
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50849004
Cr-Commit-Position: refs/heads/master@{#9142}
Passed building isac_neon and modules_unittests on Android ARMv7.
Passed modules_unittests with following filters:
--gtest_filter=FiltersTest*
--gtest_filter=LpcMaskingModelTest*
--gtest_filter=TransformTest*
--gtest_filter=FilterBanksTest*
WebRtcIsacfix_CalculateResidualEnergyNeon is removed, refer more in
Issue 4224.
The old review url is at: https://webrtc-codereview.appspot.com/37259004/
BUG=4002
R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48319005
Patch from Zhongwei Yao <zhongwei.yao@arm.com>.
Change-Id: I4c16e15930f1b3449d67b67bf023fac28121dff8
Cr-Commit-Position: refs/heads/master@{#9140}
This CL makes two changes to rtc::Buffer that have had to wait for
Chromium's use of it to be modernized:
1. Change default return type of rtc::Buffer::data() from char* to
uint8_t*. uint8_t is a more natural type for bytes, and won't
accidentally convert to a string. (Chromium previously expected
the default return type to be char, which is why
rtc::Buffer::data() initially got char as default return type in
9478437f, but that's been fixed now.)
2. Stop accepting void* inputs in constructors and methods. While
this is convenient, it's also dangerous since any pointer type
will implicitly convert to void*.
(This was previously committed (9e1a6d7c) but had to be reverted
(cbf09274) because Chromium on Android wasn't quite ready for it).
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47109004
Cr-Commit-Position: refs/heads/master@{#9132}
This is the second try at this. The previous try (a8e285d1) had to be
reverted (bd67f66e) because Chromium still declared a dependency on
move.h.
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48319004
Cr-Commit-Position: refs/heads/master@{#9123}
This CL makes two changes to rtc::Buffer that have had to wait for
Chromium's use of it to be modernized:
1. Change default return type of rtc::Buffer::data() from char* to
uint8_t*. uint8_t is a more natural type for bytes, and won't
accidentally convert to a string. (Chromium previously expected
the default return type to be char, which is why
rtc::Buffer::data() initially got char as default return type in
9478437f, but that's been fixed now.)
2. Stop accepting void* inputs in constructors and methods. While
this is convenient, it's also dangerous since any pointer type
will implicitly convert to void*.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44269004
Cr-Commit-Position: refs/heads/master@{#9121}
Removes FixedSizeLockFreeQueue which isn't used anymore. This enabled
moving rtc::AtomicOps to webrtc/base/atomicops.h where they should be.
BUG=4330
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51789004
Cr-Commit-Position: refs/heads/master@{#9120}
Instead a ViECapturer object is allocated and directly operated on. This
additionally exposes ViESharedData to Call to access the module
ProcessThread, moving towards Call ownership of shared resources.
BUG=1695
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45339004
Cr-Commit-Position: refs/heads/master@{#9119}
Changes:
1. Added MockTransport.
2. Added VoiceEngineFixture class to be used in VoE unit tests. This class is in no way final as it is - it will grow as we add more unit tests and our understanding of the Voice Engine grows.
3. Modified VoEBase unit test to use VoiceEngineFixture.
4. Added VoENetwork unit test.
R=henrika@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49759004
Cr-Commit-Position: refs/heads/master@{#9118}
In AEC a fixed fft size is used, but processing can in the lower band be in either 8 or 16 kHz.
Therefore we need a multiplier/rate factor to, for example, map frequency bands in Hz to frequency bins.
The multiplier/rate factor can only be either 1 or 2, but when 48 kHz support was added it was assigned 3.
BUG=crbug.com/482424
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43329004
Cr-Commit-Position: refs/heads/master@{#9117}
Replaces interface usage with direct calls on ViEEncoder removing a
layer of indirection. Also removing some methods from ViEImageProcess
that were only added for Video{Send,Receive}Stream usage.
BUG=1695
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45319004
Cr-Commit-Position: refs/heads/master@{#9111}
This pattern (read-only base, writable subclass) was picked to maintain a *Buffer option that doesn't copy the source bits when parsing. ByteBuffer and Buffer both copy. I'm open to discussion on what the type relationship would be, though :)
Tests have been added to ensure the symmetric nature of read/write operations.
BUG=
R=bcornell@google.com, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45259005
Cr-Commit-Position: refs/heads/master@{#9107}
Add a propagation delay to tests and make the run-time configurable for the fairness tests.
Handle losses in-between feedback messages.
BUG=4549
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49819004
Cr-Commit-Position: refs/heads/master@{#9099}
The way SetExtraOptions() is used today only applies for any one configuration change. The correct way is to set it after all flags have been scanned.
The prefered way to solve this is to use gflags and scan once, followed by applying the configuration when creating audio_processing. This is what is done in the new test tool audioproc_float.cc, but there are still some things left to do before we can replace this one.
BUG=N/A
TESTED=locally
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45279004
Cr-Commit-Position: refs/heads/master@{#9097}
ViECapturer::OnIncomingCapturedFrame can abort if two frames without a
timestamp are delivered at the same time (the same NTP timestamp is
generated). This currently causes captured_frame_ to be in a state
where timestamps (NTP time etc.) are set, but RTP timestamp is not. This
is an issue if the previous frame in captured_frame_ has yet to be
picked up by the capturer thread, as it will pick up captured_frame_ in
a state where the RTP timestamp has not been set.
BUG=chromium:480953
R=magjed@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45299004
Cr-Commit-Position: refs/heads/master@{#9094}
Passed building isac_neon and modules_unittests on Android ARM64 and
ARMv7.
Passed modules_unittests with following filters:
--gtest_filter=FiltersTest*
--gtest_filter=LpcMaskingModelTest*
--gtest_filter=TransformTest*
--gtest_filter=FilterBanksTest*
WebRtcIsacfix_CalculateResidualEnergyNeon is not enabled due to Issue
4224.
BUG=4002
R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44229004
Patch from Zhongwei Yao <zhongwei.yao@arm.com>.
Cr-Commit-Position: refs/heads/master@{#9092}
The implementation is a FIR filter bank with DCT modulation, similar to the proposed in "Multirate Signal Processing for Communication Systems" by Fredric J Harris.
The lowpass filter prototype has these characteristics:
* Passband ripple = 0.3dB
* Passband frequency = 0.147 (7kHz at 48kHz)
* Stopband attenuation = 40dB
* Stopband frequency = 0.192 (9.2kHz at 48kHz)
* Delay = 24 samples (500us at 48kHz)
* Linear phase
This filter bank does not satisfy perfect reconstruction. The SNR after analysis and synthesis (with no processing in between) is approximately 9.5dB depending on the input signal after compensating for the delay.
The performance on my workstation of AudioProcessing (with AGC and NS enabled) on a 413s recording compared to previous versions is as follows:
* Input signal has 32kHz sample rate: 3.01s
* Resampling 48kHz to 32kHz: 3.56s
* Today's temporary filter bank: 5.67s
* This filter-bank: 4.62s
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48999005
Cr-Commit-Position: refs/heads/master@{#9090}
This class is being used from both libjingle and webrtc but we recently had a regression when we added dependency on libjingle's Thread class. This cleans that up and moves the implementation of CriticalSection and helper classes into the source file.
I'm also improving debugging facilities and constness.
BUG=
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51769004
Cr-Commit-Position: refs/heads/master@{#9089}
This makes the build more flexible when linking against
prebuilt external libraries.
Use existing build_* variables for libyuv and json in talk/
(already in use in webrtc/).
Also make it possible to avoid building the GTK parts of the Linux build.
BUG=4242
R=andrew@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44179005
Cr-Commit-Position: refs/heads/master@{#9087}
