(This reverts commit 3fbf3f8841b5460503fb646eaedcb063620434a8.)
The original submission was reverted because it broke the Chrome build. This is fixed in patch set 2 of this change by keeping the old MediaConstraintsInterface string kExperimentalEchoCancellation. It will be removed once the Chrome code has been updated.
Original description:
"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=webrtc:4696
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1151573021.
Cr-Commit-Position: refs/heads/master@{#9401}
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}
When initiating a call default audio options are applied, which turns on platform-AEC if such exists. Then, if delay agnostic AEC (DA-AEC) is enabled through a media constraint no action with respect to platform-AEC is taken (a bug) and turning on SW AEC. Hence, we run both AECs.
This CL makes sure the platform-AEC is disabled if we want to run DA-AEC.
BUG=
TESTED=locally on Nexus 4 and Nexus 6.
R=henrika@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52049004
Cr-Commit-Position: refs/heads/master@{#9361}
This fixed the warning:
app/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java:46: warning: [deprecation] Assert in junit.framework has been deprecated
import static junit.framework.Assert.*;
R=glaznev@webrtc.org, pthatcher@webrtc.org
BUG=none
Review URL: https://webrtc-codereview.appspot.com/50209004
Cr-Commit-Position: refs/heads/master@{#9356}
This change connects currentAccelerateRate and currentPreemptiveRate
in webrtc::NetworkStatistics, through corresponding variables in
VoiceReceiverInfo, to googAccelerateRate and googPreemptiveExpandRate.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50179004
Cr-Commit-Position: refs/heads/master@{#9350}
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}
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}
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}
This avoids a deadlock in WebRtcVideoCapturer.
The deadlock could occur because OnIncomingFrame() has the |critical_section_stopping_| lock, which could block a Stop() on the |start_thread_|. When OnIncomingFrame() then tries to do synchronous invoke on |start_thread_| (before releasing said lock) we have a deadlock.
BUG=4670
R=magjed@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47259004
Cr-Commit-Position: refs/heads/master@{#9294}
BUG=4690
I have removed methods in VoE interfaces that were marked to be removed. I have removed them also in fake and mock implementations. I have also updated the callers in various ways:
1. Project win_test had some calls to the removed methods, but it turned out that the project is not used anymore, so I removed it entirely.
2. There were some calls to removed methods in jni methods. I have removed couple of jni methods as now they seem to do nothing.
3. With the remaining callers I just removed the calls to removed methods.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53519004
Cr-Commit-Position: refs/heads/master@{#9281}
This is being done in preparation of moving base/logging.* to rtc_base_approved. base/stream.* has libjingle dependencies that webrtc can't use, so logging.* can't depend on streams. It does look like stream.* isn't used much, so cleaning that up as well as cleaning up usage of the actual stream support (now LogStream) in the logging code, is in order, but I'll leave that to another cl.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54529004
Cr-Commit-Position: refs/heads/master@{#9269}
I wanted to use Config::Get in Chromium code, but it triggered the following
warning:
../../third_party/webrtc/common.h:89:20: error: declaration requires an exit-time destructor [-Werror,-Wexit-time-destructors]
static const T def;
^
../../third_party/webrtc/common.h:110:10: note: in instantiation of function template specialization requested here
return default_value<T>();
^
I assume we don't hit this in webrtc because the warning is disabled.
This also switches to the RTC_ prefix from the deprecated LIBJINGLE_.
Needed due to this Chromium CL:
https://codereview.chromium.org/1148843004/R=andresp@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53459004
Cr-Commit-Position: refs/heads/master@{#9268}
This fixes a bug where, if the VideoCapturer failed to start under certain circumstances, the capture manager would cause a callback saying that the capturer stopped even though it never started in the first place. A VERIFY check in VideoSource::SetState would then cause a crash since the state was set to kEnded when it was already in state kEnded (SetState only allows being called when the state changes).
I only noticed this bug while doing a mistake in a separate CL. Not sure how to reliably reproduce said bug on a working build, but I have previously had camera hardware issues where it couldn't start the camera which resulted in the SetState kEnded -> kEnded crash. Hopefully this will fix that.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51039004
Cr-Commit-Position: refs/heads/master@{#9259}
