Verifies that reduced-size isn't configured in WebRtcVideoEngine2
without explicit configuration (which doesn't exist). Also disables REMB
in the default config because it requires reconfiguration.
Adds default-config tests to make sure that they don't contain
parameters that need to be negotiated between clients.
BUG=chromium:497103, webrtc:4745
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1171533002
Cr-Commit-Position: refs/heads/master@{#9384}
We use this Config struct for enabling/disabling Extended filter mode
in AEC. This change renames it to ExtendedFilter for readability
reasons. The corresponding media constraint is also renamed to
kExtendedFilterEchoCancellation.
The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, if any of the two Configs are enabled, the extended filter
mode is engaged in APM. That is, the two Configs are combined with an
"OR" operation.
This change also renames experimental_aec in AudioOptions to extended_filter_aec.
BUG=4696
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54659004
Cr-Commit-Position: refs/heads/master@{#9378}
The delay agnostic AEC uses a signal based delay correction method to adjust buffer synchronization between loudspeaker and microphone. On Mac in particular we have seen deviations in UMA stats that point towards an echo already at startup. This is likely due to an early and incorrect correction based on poor audio data.
By waiting 15 seconds before we turn on the ability to correct we can avoid a majority of these.
The reported delay values are in general accurate enough and relying on them in the beginning is fine. The value 15 seconds is chosen because we have seen from UMA data that a significant amount of calls tend to end before 15 seconds when being in the UseDelayAgnosticAEC Finch experiment.
We turn this "feature" on for all platforms but Android, where the reported system delays are inaccurate and we want to take action as soon as possible.
In addition, the set of "good" delay values has been increased from 25% to 75% of the filter length.
BUG=webrtc:3504
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50199004
Cr-Commit-Position: refs/heads/master@{#9376}
The existing style in these files is pretty inconsistent and wildly divergent
from most of WebRTC/Chromium; clang-formatting them not only makes them easier
to read, it makes me see fewer presubmit errors when I try to touch the files to
make other changes.
BUG=none
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52019004
Cr-Commit-Position: refs/heads/master@{#9364}
This reverts commit d8a03facf6986a011c8f889c63d87f9216a1e912, since it
broke the Chrome build. Will have to swap to using base/logging.h in
neteq_impl.cc before re-landing this change.
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50219004
Cr-Commit-Position: refs/heads/master@{#9360}
This was not implemented before. It returns the current total delay (packet buffer and sync buffer) of NetEq. This is the same information that was already available in NetEqNetworkStatistics::current_buffer_size_ms, that can be obtained through NetEq::NetworkStatistics(). But, since the current delay is a key metric of NetEq, it is convenient to have it available in a simpler way.
R=kwiberg@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51149004
Cr-Commit-Position: refs/heads/master@{#9359}
This CL connects RTCConfiguration::audioJitterBufferFastMode in
PeerConnection.java, through libjingle, down to
NetEq::Config::enable_fast_accelerate in native WebRTC.
When enabled, it will allow NetEq to do faster time-compression when
the buffer level is very high.
BUG=4691
R=henrika@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55479004
Cr-Commit-Position: refs/heads/master@{#9344}
A modified operation mode was added, holding:
--- Stricter conditions for AcceleratedRampUp.
--- Smoother GradualRateUpdate adjustments.
--- New AcceleratedRampDown update mode.
This mode reduces significantly the delay for bitrates around its minimum bound.
Several NADA unittests and a few simulations were added.
Fixed LinkedSet bug.
Fixed IsNewerSequenceNumber/IsNewerTimestamp bug.
BUG=4550
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54399004
Cr-Commit-Position: refs/heads/master@{#9340}
BUG=4690
Changes:
1. In MediaEngineInterface changed CreateChannel() to CreateChannel(const AudioOptions&). Plan is to eventually remove Get/SetAudioOptions and the cousins SetDelayOffset and SetDevices.
2. In ChannelManager changed CreateVoiceChannel(...) to CreateVoiceChannel(..., const AudioOptions&).
3. In ChannelManager removed SetEngineAudioOptions, because it is not used and we want to eventually remove SetAudioOptions.
4. Updated MediaEngineInterface implementations and unit tests accordingly.
5. In WebRtcVoiceEngine changed access of Set/ClearOptionOverrides to protected. These are only used by WebRtcVoiceMediaChannel (now a friend). Plan is to rethink the logic behind option overrides.
6. Cosmetics: replaced NULL with nullptr in touched code
R=solenberg@google.com, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/56499004
Cr-Commit-Position: refs/heads/master@{#9330}
The old OpenSSL threading hooks were removed in favor of the library knowing
about threads internally. Instead of CRYPTO_add, use FOO_up_ref wrappers that
don't require reaching into the type.
BUG=none
R=jiayl@webrtc.org, juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/54579004
Cr-Commit-Position: refs/heads/master@{#9324}
Use the current parameter names in the comment for SetNackMode().
Add a warning comment about the lifetime of the return value of
GetNackList().
R=stefan@webrtc.org
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/52599004
Cr-Commit-Position: refs/heads/master@{#9321}
The most important change is to prevent a potential buffer overflow in
NackList(). It cannot happen if the |size| argument passed to NackList()
is consistent with the |max_nack_list_size| argument passed to
SetNackSettings(), and there is an assertion to check that. But it is
good to defend against this in the release build because assert() is
compiled away in the release build.
Remove the unused |master| parameter to the VCMReceiver constructor.
Remove the unused State() getter method and the corresponding state_
member.
Remove the declarations for the nonexistent GenerateReceiverId()
method and the receiver_id_counter_ member.
Remove the unneeded data_buffer_ member of TestVCMReceiver. It was
assigned to packet.dataPtr and then immediately overwritten by
stream_generator_->GetPacket() or stream_generator_->PopPacket().
R=stefan@webrtc.org
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/51119004
Cr-Commit-Position: refs/heads/master@{#9318}
Reduces musical noise with minimal impact on interferer suppression.
This also unifies the treatment of "mean bins". The "end" bin is now
inclusive in the mean range as with the "start" bin.
Corrects interpretation of quantile.
BUG=chromium:490477
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50939004
Cr-Commit-Position: refs/heads/master@{#9317}
Changed Create() to return a pointer to the object rather than an error message, which is in line with how objects should be created.
BUG=441
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51939004
Cr-Commit-Position: refs/heads/master@{#9315}
