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}
Registers transport on construction removing the need for ViESender as a
hop and removing a potential deadlock by removing RegisterSendTransport.
BUG=1695, 2999
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/57449004
Cr-Commit-Position: refs/heads/master@{#9309}
Commit 7e0c7d49 ("Add support for external encoders in ACM") changed
things around so that we no longer recreate the speech encoder when
adding CNG or RED to an existing encoder. This isn't correct, since
those two expect to be in sync with the speech encoder they work with.
Solve the problem by resetting the speech encoder before hooking in
RED or CNG.
BUG=crbug/490368
R=jmarusic@webrtc.orgTBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53589004
Cr-Commit-Position: refs/heads/master@{#9307}
The purpose of the tool is to analyze the output from the command line
tool rtp_analyze. That is, starting with an rtpdump or pcap file, it
is processed through rtp_analyze to produce a text output, which is
then used as input to this new Matlab function.
BUG=2692
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47339004
Cr-Commit-Position: refs/heads/master@{#9306}
If srtp_create fails while adding streams, it deallocates the session
but doesn't clear the passed pointer which then could lead to a
double-free in the SrtpSession dtor.
The CL also adds locking for libsrtp initialization / shutdown.
BUG=4042
R=jiayl@webrtc.org, juberti@google.com, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47319004
Cr-Commit-Position: refs/heads/master@{#9300}
