After offline discussion, this was mostly of historical use and may
actually be more of a risk now that we have the paced sender.
BUG=
Review-Url: https://codereview.webrtc.org/2198933004
Cr-Commit-Position: refs/heads/master@{#13643}
Also lowering the min bitrate for simulations to be able to better capture this issue in the BweFeedbackTest.Choke200kbps30kbps200kbps performance test.
BUG=webrtc:6105
NOTRY=true
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2201093006
Cr-Commit-Position: refs/heads/master@{#13639}
This is part of rewriting the ConferenceMixer and OutputMixer.
Calls are instead routed through AudioReceiveStream::Start/Stop.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2206223002
Cr-Commit-Position: refs/heads/master@{#13636}
GCM cipher suites are optional (disabled by default) and can be enabled
through "PeerConnectionFactoryInterface::Options".
If compiled with Chromium (i.e. "ENABLE_EXTERNAL_AUTH" is defined), no
GCM ciphers can be used yet (see https://crbug.com/628400).
BUG=webrtc:5222, 628400
Review-Url: https://codereview.webrtc.org/1528843005
Cr-Commit-Position: refs/heads/master@{#13635}
Add an ICE nomination attribute. If a connection switched on the controlling side, increase the nomination value set in the attribute.
The controlled side will also be ready for re-nomination option; it will switch if a nomination comes with a higher nomination value even though it may be at a lower priority.
Plus, don't nominate or re-nominate if the nomination value at the current connection has been acknowledged.
BUG=
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2163403002 .
Cr-Commit-Position: refs/heads/master@{#13631}
Scaling causes us to work the CPU too much, which very quickly degrades quality. This causes us to at least behave better on good networks.
NOTRY=True
BUG=
Review-Url: https://codereview.webrtc.org/2205763002
Cr-Commit-Position: refs/heads/master@{#13630}
Additional changes I needed to make it work:
- Modified a header in RTPFile.cc. Every other file is
using "webrtc/engine_configurations.h" instead.
- Disabled flag 4373 for msvs because it was disabled
in build/common.gypi.
BUG=webrtc:6038
TBR=kwiberg@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2187563005
Cr-Commit-Position: refs/heads/master@{#13628}
- Removes unnecessary casts to compute timedelta.
- Renames ProbingState for clarity. This should help when we probe mid-call.
- Enables probing by default to avoid checking on each incoming packet.
- Removes duplicate probing state tracking in paced sender. These duplicate
states were conflicting at times.
- Removes passing through packets for bug 5307 which seems long fixed.
- Cleanup handling of time_last_send_ms and avoid side effects of changing
probing state at TimeUntilNextProbe().
- Clear cluster data when probing is restarted to avoid having old data after a reset.
BUG=5859
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2182603002 .
Cr-Commit-Position: refs/heads/master@{#13612}
It's not part of the gyp build either, and `ninja -C out/gnios webrtc`
causes several build errors, at least in Chromium builds (which means
gn/ios bots building 'all' are broken).
BUG=chromium:633316
R=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/2205533003 .
Cr-Commit-Position: refs/heads/master@{#13609}
It was possible that "A < B" and "B < A" both evaluated to true.
This manifested as an assert on Windows, and a memory leak on Linux.
Note that the concept of "less than" is meaningless for this object.
The operator is only needed so the object can be used as a key in an
std::map.
BUG=webrtc:6068
R=honghaiz@webrtc.org, kjellander@webrtc.org, skvlad@webrtc.org
Review URL: https://codereview.webrtc.org/2187913002 .
Cr-Commit-Position: refs/heads/master@{#13598}
This test verifies that SRTP errors are only signaled if a specific
amount of time has passed, and was using ProcessMessages(time) to
wait for an amount of time. But ProcessMessages may sometimes wait
for longer than the requested time (especially on TSAN bot, etc.).
BUG=webrtc:4690
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2184083004 .
Cr-Commit-Position: refs/heads/master@{#13597}
Passing transport_frame_id() to VideoSink will allow to identify incoming video
frames, which will make it possible to correlate video frames on the
sender and on the receiver.
BUG=chromium:621691
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2088953002 .
Cr-Commit-Position: refs/heads/master@{#13596}
A reset of Bwe and rates on network route change triggers
switching of the RemoteBitrateEstimator from DelayBasedBwe
to RemoteBitrateEstimatorAbsSendTime. This is unexpected,
fix this.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2193043002 .
Cr-Commit-Position: refs/heads/master@{#13594}
Also makes the properties protected, as previously requested by Stefan.
Review-Url: https://codereview.webrtc.org/2179223003
Cr-Commit-Position: refs/heads/master@{#13593}
The intended behavior of the pacer is that it should send audio (i.e high priority) packets immediatelly even if the pacer is paused and even if the available media budget has been used. The existing code will not send audio if the pacing budget has been used. It will normally send audio when the pacer is paused, but if the media budget was already used up when the pacer was paused, it would neither send audio nor update the budget.
Updated the implementatation and the unit test. Added logging for when the pacer is paused, and when it receives a zero bitrate estimate.
BUG=webrtc:6155
Review-Url: https://codereview.webrtc.org/2193183003
Cr-Commit-Position: refs/heads/master@{#13592}
The ProbingCalculator class calculates and validates the results from
probing attempts.
BUG=webrtc:5859
Review-Url: https://codereview.webrtc.org/2121183002
Cr-Commit-Position: refs/heads/master@{#13589}
