With this change, the aggregates for packet waiting times are
calculated in NetEq's StatisticsCalculator insead of in
AcmReceiver. This simplifies things somewhat, and avoids having to
copy the raw data on polling.
R=ivoc@webrtc.org, minyue@webrtc.org
Review URL: https://codereview.webrtc.org/1296633002 .
Cr-Commit-Position: refs/heads/master@{#9778}
By converting three raw pointers to scoped_ptrs, we can eliminate the
need for a manually-defined destructor, and generally sleep better at
night.
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1310213003 .
Cr-Commit-Position: refs/heads/master@{#9776}
This CL allows you to, having generated one or more RTCCertificates, supply them to RTCConfiguration for CreatePeerConnection use. This means an SSLIdentity does not have to be generated with a DtlsIdentityStore[Interface/Impl] as part of the CreatePeerConnection steps because the certificate contains all the necessary information.
To create an RTCCertificate you have to do the identity generation yourself though. But you could reuse the same RTCCertificate for multiple connections.
BUG=webrtc:4927
R=tommi@webrtc.org, torbjorng@webrtc.org
Review URL: https://codereview.webrtc.org/1288033009 .
Cr-Commit-Position: refs/heads/master@{#9774}
During the reconnection phase, EWOULDBLOCK has been returned to upper layer which stops the sending of video stream.
BUG=webrtc:4930
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1288553010 .
Cr-Commit-Position: refs/heads/master@{#9767}
The only shared state is now the bandwidth estimation info.
This reduces the amount and complexity of the locking
substantially.
Review URL: https://codereview.webrtc.org/1208993010
Cr-Commit-Position: refs/heads/master@{#9762}
AudioDeviceTemplate doesn't initialize `output_` and `input_` if the
initialization of `audio_manager_` succeeds. Similarly, it doesn't
terminate `input_` and `audio_manager_` if the termination of `output_`
succeeds. This CL fixes this.
BUG=
Review URL: https://codereview.webrtc.org/1296693003
Cr-Commit-Position: refs/heads/master@{#9760}
This functionality is not used internally in WebRTC. Also, it's not safe, because the frame is supposed to be read-only, and it will likely not work for texture frames.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1296113002 .
Cr-Commit-Position: refs/heads/master@{#9753}
This makes us use the clever code that adapts the path depending on where we build, so it works in google3 too.
BUG=None
Review URL: https://codereview.webrtc.org/1291283003
Cr-Commit-Position: refs/heads/master@{#9752}
If the same extension URI is used for both audio and video (such as
abs-send-time), we should be able to re-use the same ID. A conflict
only exists if two different URIs are attempting to use the same ID.
Review URL: https://codereview.webrtc.org/1286273003
Cr-Commit-Position: refs/heads/master@{#9749}
Adding 'ReceiveCodecsHaveChanged' method that will determine if codecs
HAVE changed, irrespective of order and preference.
Review URL: https://codereview.webrtc.org/1291763003
Cr-Commit-Position: refs/heads/master@{#9748}
On the controlled side, a stun request without use-candidate attribute will
be used for sending media.
BUG=4900
Review URL: https://codereview.webrtc.org/1270613006
Cr-Commit-Position: refs/heads/master@{#9747}
Migrated from https://codereview.webrtc.org/1275703006/ which causes test failures for android. On android, loopback interface was used as local interface to generate candidates. Add a test case to make sure this won't be broken in the future.
Also observed some failures under content_browsertests in chromium.fyi bot but can't repro locally. Might just be temporary test issue.
BUG=webrtc:4517
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1299333003 .
Cr-Commit-Position: refs/heads/master@{#9746}
This cl removes a lot of unused functionality in SystemInfo and reduces dependencies.
There's duplicate functionality in this class and WebRTC's CpuInfo class, so I'm consolidating the two implementations.
The CpuInfo interface will still exist since it's being used from Chrome to initialize the core count before the sandbox is engaged.
With this change, the SystemInfo class will get the benefit too of this initialization.
NOTRY=true
(using no try due to a dead android try bot)
Review URL: https://codereview.webrtc.org/1286163003
Cr-Commit-Position: refs/heads/master@{#9743}
Negative values from TimeUntilNextProcess indicate that the module
wanted to run sooner than possible, not that an invalid error code was
returned. As such it's not a contract error.
BUG=webrtc:4879
NOTRY=true
Review URL: https://codereview.webrtc.org/1257833004
Cr-Commit-Position: refs/heads/master@{#9740}
The IceEndpointType has the format of <local_endpoint>_<remote_endpoint>. It is recorded on the BestConnection when we have the first OnTransportCompleted signaled.
BUG=webrtc:4918
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1277263002 .
Cr-Commit-Position: refs/heads/master@{#9737}
Adapter enumeration disable should be a concept from Network. Network will be hooked up with media permission (mic/camera) to check whether gathering local address is allowed.
BUG=crbug.com/520101
R=juberti@webrtc.org, pthatcher@webrtc.org
Committed: ba9ab4cd8d
Review URL: https://codereview.webrtc.org/1284113003 .
Cr-Commit-Position: refs/heads/master@{#9735}
This is more a formality than anything since scoped_ptr.h is already being included from rtc_base_approved code.
BUG=
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1287203007 .
Cr-Commit-Position: refs/heads/master@{#9732}
This CL adds calculation and logging of average excess buffer delay
and number of delayed packet outage events per minute.
The first is the average of time spent in the packet buffer for all
packets. The average is calculated for intervals of one minute, and
the result is logged to the UMA stat
WebRTC.Audio.AverageExcessBufferDelayMs.
The second is a counter of delayed packet outage events that is
restarted every minute, and the result is logged to the UMA stat
WebRTC.Audio.DelayedPacketOutageEventsPerMinute. For a description of
delayed packet outages, see previous CL implementing a duration log
for these events.
BUG=webrtc:4915, chromium:488124
R=minyue@webrtc.org
Review URL: https://codereview.webrtc.org/1287333005 .
Cr-Commit-Position: refs/heads/master@{#9731}
This reverts commit 0a2955f227666efd87b2a303a69c083ef801c528.
Revert "In the past, P2PPortAllocator.enable_multiple_routes is the indicator whether we should bind to the any address. It's easy to translate that into a port allocator flag in P2PPortAllocator's ctor. Going forward, we have to depend on an asynchronous permission check to determine whether gathering local address is allowed or not, hence the current way of passing it through constructor approach won't work any more. The asynchronous check will trigger SignalNetowrksChanged so we could only check that inside DoAllocate."
This reverts commit ba9ab4cd8d2e8fbc068dc36b5e6f6331d7deeccf.
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1288843003 .
Cr-Commit-Position: refs/heads/master@{#9729}
