However, two other "hacks" had to be added to maintain bit-exactness
with legacy.
Note that this change requires a new version of the universal.rtp test
input, although the output reference stays the same.
Moving reference files, and using a new input vector for NetEq4.
The new input vector neteq_universal_new.rtp is identical to the old
neteq_universal.rtp, except that the payload type for CNG packets that
follows a wideband codec is changed to 98.
Update to resources revision 15 where the new reference files are.
Also changing a faulty log error.
Review URL: https://webrtc-codereview.appspot.com/1078009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3442 4adac7df-926f-26a2-2b94-8c16560cd09d
> Delay estimator wrapper API changes. This should finalize the changes to delay estimator making it work for multi-probe.
>
> The changes are summarized here:
>
> delay_estimator.*
> -----------------
> Replaced assert() with correct error check. This is consistent with previous versions of the delay_estimator, i.e., to check for valid parameters where they are actually used and not high up in a wrapper layer.
>
> delay_estimator_internal.h
> --------------------------
> Pulled out the far-end part of DelayEstimator struct and put it in DelayEstimatorFarend. The only common parameter is spectrum_size, which we store in both and thereby avoiding having a Farend pointer in DelayEstimator.
>
> delay_estimator_wrapper.*
> -------------------------
> Added and updated descriptions. From Free(), Create(), Init() the far-end parts have been put in separate Farend versions. Same goes for the Process() which now has an AddFarSpectrum() version.
> The flow of calls should be something like (in pseudo-code)
>
> far* = CreateFarend(history_size)
> near* = Create(far, lookahead)
> InitFarend(far)
> Init(near)
> while call ongoing
> AddFarSpectrum(far, far_spectrum)
> Process(near, near_spectrum)
> end while
> Free(near)
> FreeFarend(far)
>
> delay_estimator_unittest.cc
> ---------------------------
> Added farend support setting up calls as mentioned above.
>
> aecm_core.*
> -----------
> Cleaned up some lint warnings.
> Added delay_estimator_farend pointer. Called Create(), Init() and Free() in above mentioned order.
> If AddFarSpectrumFix() was not successfully done, we end and return -1. This is what we would have done for Process().
>
> aec_core.*
> ----------
> Cleaned up some lint warnings.
> Added delay_estimator_farend pointer. Calls in proper order. Since we only use the delay estimator for logging there is no error handling. We only call Process() if AddFarSpectrum() was successful though.
>
> TEST=audioproc_unittest, trybots
> BUG=None
>
> Review URL: https://webrtc-codereview.appspot.com/1076006TBR=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1062008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3429 4adac7df-926f-26a2-2b94-8c16560cd09d
The changes are summarized here:
delay_estimator.*
-----------------
Replaced assert() with correct error check. This is consistent with previous versions of the delay_estimator, i.e., to check for valid parameters where they are actually used and not high up in a wrapper layer.
delay_estimator_internal.h
--------------------------
Pulled out the far-end part of DelayEstimator struct and put it in DelayEstimatorFarend. The only common parameter is spectrum_size, which we store in both and thereby avoiding having a Farend pointer in DelayEstimator.
delay_estimator_wrapper.*
-------------------------
Added and updated descriptions. From Free(), Create(), Init() the far-end parts have been put in separate Farend versions. Same goes for the Process() which now has an AddFarSpectrum() version.
The flow of calls should be something like (in pseudo-code)
far* = CreateFarend(history_size)
near* = Create(far, lookahead)
InitFarend(far)
Init(near)
while call ongoing
AddFarSpectrum(far, far_spectrum)
Process(near, near_spectrum)
end while
Free(near)
FreeFarend(far)
delay_estimator_unittest.cc
---------------------------
Added farend support setting up calls as mentioned above.
aecm_core.*
-----------
Cleaned up some lint warnings.
Added delay_estimator_farend pointer. Called Create(), Init() and Free() in above mentioned order.
If AddFarSpectrumFix() was not successfully done, we end and return -1. This is what we would have done for Process().
aec_core.*
----------
Cleaned up some lint warnings.
Added delay_estimator_farend pointer. Calls in proper order. Since we only use the delay estimator for logging there is no error handling. We only call Process() if AddFarSpectrum() was successful though.
TEST=audioproc_unittest, trybots
BUG=None
Review URL: https://webrtc-codereview.appspot.com/1076006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3428 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL is one step in a larger change of the DelayEstimator where we will open up for multiple near-end signals.
This particular CL separates the low level far-end parts without affecting the usage externally. This is a first step towards separating the far-end and near-end parts giving the user the control.
BUG=None
TEST=audioproc_unittests, trybots
Review URL: https://webrtc-codereview.appspot.com/1068005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3415 4adac7df-926f-26a2-2b94-8c16560cd09d
The function WebRtcOpus_DurationEst returned the number of samples
per packet in the native 48 kHz sample rate, while the decoder
function returns data in 32 kHz sample rate. This creates a discrepancy
that makes NetEQ's lip-sync functionality add too little delay.
BUG=1334
TEST=try bots
Review URL: https://webrtc-codereview.appspot.com/1069006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3403 4adac7df-926f-26a2-2b94-8c16560cd09d
Eliminated need for video receiver to talk to its parent. Also we will now determine if the packet is the first one already in the rtp general receiver. The possible downside would be that recovered video packets no longer can be flagged as the first packet, but I don't think that can happen. Even if it can happen, maybe the bit was set anyway at an earlier stage. The tests run fine.
BUG=
TEST=rtp_rtcp_unittests, vie_auto_test, voe_auto_test
Review URL: https://webrtc-codereview.appspot.com/1022011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3382 4adac7df-926f-26a2-2b94-8c16560cd09d
The goal with this new clock interface is to have something which is used all
over WebRTC to make it easier to switch clock implementation depending on where
the components are used. This is a first step in that direction.
Next steps will be to, step by step, move all modules, video engine and voice
engine over to the new interface, effectively deprecating the old clock
interfaces. Long-term my vision is that we should be able to deprecate the clock
of WebRTC and rely on the user providing the implementation.
TEST=vie_auto_test, rtp_rtcp_unittests, trybots
Review URL: https://webrtc-codereview.appspot.com/1041004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3381 4adac7df-926f-26a2-2b94-8c16560cd09d
This is a pure reformat patch, with the exception that I also fixed all comments and moved a constant. I did not change the types in this patch since I
though that is more risky, so I'll do that in a separate patch later (perhaps
we could purge the types from the whole module in one go?)
BUG=
TEST=Trybots, vie_ & voe_auto_test --automated
Review URL: https://webrtc-codereview.appspot.com/998007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3338 4adac7df-926f-26a2-2b94-8c16560cd09d
