This cl just moves the logic form the default module
SetTargetSendBitrates to PayloadRouter. There might be glitch / mismatch
in size between trate the vector and rtp modules. This was the same in
the default module and is quite hard to protect from before we have the
new video API.
I also removed some test form rtp_rtcp_impl_unittest that were affected
by this change. The test tests code that isn't implemented, hence the
DISABLED_, and this will never be implemented in the RTP module, rather
the payload router in the future.
BUG=769
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42419004
Cr-Commit-Position: refs/heads/master@{#8453}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8453 4adac7df-926f-26a2-2b94-8c16560cd09d
There is a potential race when deleting a channel and there is a frame
in the encoder. ViEEncoder::SendData can be called after
ViEEncoder::StopThreadsAndRemovePayloadRouter and payload_router is
then already removed.
Until we have the new API in place, use scoped_refptr in ViEChannel and
ViEEncoder and deregister channel/encoder before deleting.
BUG=769
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42019004
Cr-Commit-Position: refs/heads/master@{#8443}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8443 4adac7df-926f-26a2-2b94-8c16560cd09d
* Don't use ConjugateDotProduct to calculate the norm.
* Only resize Matrix when needed.
This makes the Beamformer run in 93.6% the original time.
The error between the new and original output is really small and is caused by the new norm calculation.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37339004
Cr-Commit-Position: refs/heads/master@{#8438}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8438 4adac7df-926f-26a2-2b94-8c16560cd09d
Fetching the current codec for sake of gathering stats, is frequently blocked since it's done by acquiring the same lock as is held while encoding frames. This can mean tens of milliseconds.
To improve this, I'm taking advantage of the fact that the codec information is set on the same thread as is used to query the information. This means that locking isn't needed for querying this information. I'm adding checks to make sure debug builds will crash if this isn't followed.
An alternative to this approach could be to add one more lock that is specifically used for the codec information variable. This would also decouple querying codec information from the encoder itself, but still requires a lock.
This patch depends on making ThreadChecker part of rtc_base_approved:
https://webrtc-codereview.appspot.com/40539004/
BUG=2822
R=mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37779004
Cr-Commit-Position: refs/heads/master@{#8435}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8435 4adac7df-926f-26a2-2b94-8c16560cd09d
When an "empty frame" was produced by ACMGenericCodecWrapper::Encode,
the timestamp value was not set. This is now fixed, and the first byte
of the bitstream is set to something as well to avoid similar problems.
BUG=chromium:459483
R=jmarusic@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34269004
Cr-Commit-Position: refs/heads/master@{#8431}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8431 4adac7df-926f-26a2-2b94-8c16560cd09d
WebRTC binds to individual NICs and listens for incoming Stun packets. Sending stun through this specific NIC binding could make OS route the packet differently hence exposing non-VPN public IP.
The fix here is
1. to bind to any address (0:0:0:0) instead. This way, the routing will be the same as how chrome/http is.
2. also, remove the any all 0s addresses which happens when we bind to all 0s.
BUG=4276
R=juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8418
Review URL: https://webrtc-codereview.appspot.com/39129004
Cr-Commit-Position: refs/heads/master@{#8419}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8419 4adac7df-926f-26a2-2b94-8c16560cd09d
WebRTC binds to individual NICs and listens for incoming Stun packets. Sending stun through this specific NIC binding could make OS route the packet differently hence exposing non-VPN public IP.
The fix here is
1. to bind to any address (0:0:0:0) instead. This way, the routing will be the same as how chrome/http is.
2. also, remove the any all 0s addresses which happens when we bind to all 0s.
BUG=4276
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39129004
Cr-Commit-Position: refs/heads/master@{#8418}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8418 4adac7df-926f-26a2-2b94-8c16560cd09d
* Only compare actual pixel data, not the padding between width and stride.
* When creating a frame from raw buffers with excessive size, do not assume that the frame’s allocated size will be as excessive as the input buffers.
* The arrays in TestI420VideoFrame.CopyFrame and TestI420VideoFrame.CloneFrame are too small, and we currently memcpy out of bounds.
I think this CL should land regardless, but the main purpose is to pave the way for for planned changes to I420VideoFrame. See https://review.webrtc.org/38879004.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34199004
Cr-Commit-Position: refs/heads/master@{#8416}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8416 4adac7df-926f-26a2-2b94-8c16560cd09d
The old AcmOpusTest depends on the ACMOpus class, but this class was
obsoleted by AudioEncoderOpus. In this CL, the test code is re-written
to use AudioEncoderOpus and ACMGenericCodecWrapper instead of
ACMOpus. Most of the test functionality is preserved, except for the
packet loss rate tests, which where already transferred to
AudioEncoderOpusTest in r8244.
R=kwiberg@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40029004
Cr-Commit-Position: refs/heads/master@{#8410}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8410 4adac7df-926f-26a2-2b94-8c16560cd09d
Although sample_rate_hz(), num_channels(), and rtp_timestamp_rate_hz()
are simple accessors for almost all implementations of AudioEncoder,
they are virtual and not guaranteed to be just simple accessors. Thus,
it makes more sense to use the normal CamelCase naming scheme.
BUG=4235
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34239004
Cr-Commit-Position: refs/heads/master@{#8407}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8407 4adac7df-926f-26a2-2b94-8c16560cd09d
Unblocks pending threads (render thread + decoder thread) when
destroying renderers and shutting down decoders.
Speeds up SetLocalDescription significantly (10x or so) under
WebRtcVideoEngine2 but also shutdown times in ~ViEChannel and
~ViEReceiver in general.
BUG=1788
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41959004
Cr-Commit-Position: refs/heads/master@{#8387}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8387 4adac7df-926f-26a2-2b94-8c16560cd09d
Bitrate controller is used in VoiceEngine to smoothen the fraction loss
from RTCP report blocks. This CL removes the usage of the
BitrateController and calculates its own fraction loss average insted.
This introduces some duplicated code between BitrateController and
Channel, but removes processing overhead and the incorrect bandwidth
estimation numbers reported by the bitrate controller.
BUG=4310
TEST=voe_cmd_test with network simulator
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39999004
Cr-Commit-Position: refs/heads/master@{#8386}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8386 4adac7df-926f-26a2-2b94-8c16560cd09d
