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Author SHA1 Message Date
pbos@webrtc.org
4e2806d85f Remove WEBRTC_TRACE uses in video_engine/ 
Complements fixes by mflodman@.

BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6136 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 08:02:22 +00:00
stefan@webrtc.org
24bd364d3e Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels. 
This fixes an issue where the user doesn't know which channels are "active" and therefore can't properly sum the estimates for all channels.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6041 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 12:35:37 +00:00
solenberg@webrtc.org
4e65602886 Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5791 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 14:32:47 +00:00
solenberg@webrtc.org
3fb8f7bbb0 Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5765 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-24 20:28:11 +00:00
pbos@webrtc.org
3349ae0cdc Implement minimum transmit bitrate. 
Utilizing minimum transmission bitrate prevents low remote bitrate
estimates (bitrate estimation dips) when encoding non-complex content
such as screenshare of a static image even though there's nothing wrong
with the link.

Requires pacing to be enabled for now, pending issue 3036.

BUG=3014
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5694 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-13 12:52:27 +00:00
jiayl@webrtc.org
9fd8d87ff5 Adds APIs for reporting pacer queuing delay. 
BUG=2775
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8959005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5621 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-27 22:32:40 +00:00
asapersson@webrtc.org
8098e07478 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). 
Add counter to RTCP sender and RTCP receiver.
Add video api GetRtcpPacketTypes().

BUG=2638
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5575 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 11:59:02 +00:00
jiayl@webrtc.org
1f64f06784 Add stats of incoming frame delays for debugging bandwidth estimation. 
BUG=crbug/338380
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5519 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-10 19:12:14 +00:00
sprang@webrtc.org
0e93257cee Add callbacks for receive channel RTP statistics 
This allows a listener to receive new statistics (byte/packet counts, etc) as it
is generated - avoiding the need to poll. This also makes handling stats from
multiple RTP streams more tractable. The change is primarily targeted at the new
video engine API.

TEST=Unit test in ReceiveStatisticsTest.
Integration tests to follow as call tests when fully wired up.

BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5416 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-23 10:00:39 +00:00
sprang@webrtc.org
54ae4ffb9e Add callbacks for receive channel RTCP statistics. 
This allows a listener to receive new statistics as it is generated - avoiding the need to poll. This also makes handling stats from multiple RTP streams more tractable.
The change is primarily targeted at the new video engine API.

TEST=Unit test in ReceiveStatisticsTest. Integration tests to follow as call tests when fully wired up.

BUG=2235
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5323 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-19 13:26:02 +00:00
pbos@webrtc.org
5ab756703e Revert r5294 to re-roll r5293. 
To fix races in test each stream now owns its own encoder/decoder.

R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/5919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5297 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-16 12:24:44 +00:00
turaj@webrtc.org
41e2615e02 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." 
> Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
> 
> BUG=
> R=mflodman@webrtc.org, stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/5409004

TBR=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5294 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-15 18:42:32 +00:00
solenberg@webrtc.org
341e91441a Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 23:57:54 +00:00
sprang@webrtc.org
6811b6e308 Callback for send bitrate estimates - new roll 
Issue https://webrtc-codereview.appspot.com/4459004/ was commited as
r5259, after which flakiness was detected and a rollback was performed
at r5261.

Patch Set 1 of this issue is the code submitted in r5259. Subsequent
patch sets fixes a race condition which caused the seen problems.

The root cause was a dead lock between a thread sending rtp packets and
and a timed module processing thread:

webrtc::RTPSender::BitrateUpdated()  // Get RTPSender stats lock
webrtc::Bitrate::Process()  // Get Bitrate lock
webrtc::RTPSender::ProcessBitrate()
webrtc::ModuleRtpRtcpImpl::Process()
...

webrtc::Bitrate::Update()  // Get Bitrate lock
webrtc::RTPSender::UpdateRtpStats()  // Get RTPSender stats lock
webrtc::RTPSender::SendToNetwork()
...

This is fixed in Bitrate::Process() by releasing the lock before
calling the callback.

BUG=2235
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5281 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 09:46:59 +00:00
sprang@webrtc.org
096e8d9f94 Revert 5259 "Callback for send bitrate estimates" 
CL is causing flakiness in RampUpTest.WithoutPacing.

> Callback for send bitrate estimates
>
> BUG=2235
> R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/4459004

R=mflodman@webrtc.org, pbos@webrtc.org
TBR=mflodman

Review URL: https://webrtc-codereview.appspot.com/5579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5261 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-11 14:07:33 +00:00
sprang@webrtc.org
2656cf9f4c Callback for send bitrate estimates 
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5259 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-11 12:53:03 +00:00
sprang@webrtc.org
7f73280ded Fraction lost statistics not being reported 
A bug is causing fraction lost to always be set to zero when calling
ViERTP_RTCP::Get(Send|Receive)ChannelRtcpStatistics. Fix this and update
tests to catch it.

BUG=
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5235 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-06 11:56:55 +00:00
sprang@webrtc.org
ebad765ee0 Add callbacks for send channel rtp statistics 
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5227 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-05 14:29:02 +00:00
sprang@webrtc.org
a6ad6e5b58 Add callbacks for send channel rtcp statistics 
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5220 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-05 09:48:44 +00:00
sprang@webrtc.org
71f055fb41 Add send frame rate statistics callback 
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4479005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5213 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-04 15:09:27 +00:00
asapersson@webrtc.org
8d02f5dc71 Added API for enabling/disabling RTCP Receiver Reference Time extension. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3419005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5147 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 08:57:04 +00:00
sprang@webrtc.org
dc50aaeaa8 Interface changes to old api, for use by new api transition. 
BUG=2589
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5142 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 16:47:07 +00:00
wu@webrtc.org
822fbd8b68 Update talk to 50918584. 
Together with Stefan's http://review.webrtc.org/1960004/.

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2048004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-15 23:38:54 +00:00
tnakamura@webrtc.org
aa4d96a134 Revert r4301 
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 19:25:04 +00:00
stefan@webrtc.org
66b2e5c05a Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the 
rtp_rtcp implementation.

This refactoring significantly reduces the receive-side RTP parser and receiver
complexity, and makes it possible to implement RTX correctly by having two
instances of receive-statistics.

With this change the dead-or-alive and packet timeout APIs are removed.

TEST=trybots, vie_auto_test, voe_auto_test
BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1745004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-05 14:30:48 +00:00
solenberg@webrtc.org
a6db54d4c9 - Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. 
- Changed implementation of SetReceiveAbsoluteSendTimeStatus API so the RBE instance is changed when at least one channel in a group has the extension enabled.

BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1553005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4113 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-27 16:02:56 +00:00
stefan@webrtc.org
c74c3c2447 Adds integration test for RTX and fixes bugs found. 
BUG=1811
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4096 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-23 13:48:22 +00:00
solenberg@webrtc.org
cb9cff0c71 Add functions to ViE API to enable/disable the absolute send time header extension. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1487004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4065 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-20 12:00:23 +00:00
pbos@webrtc.org
f5d4cb1958 Include files from webrtc/.. paths in video_engine/ 
BUG=1662
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1492004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4056 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-17 13:44:48 +00:00
mflodman@webrtc.org
4dee30927a Remove SetOverUseDetectorOptions and cleaned ViESharedData. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1486004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4042 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 11:13:18 +00:00
andresp@webrtc.org
29b2219914 Adding a factory to remote bitrate estimator and allow it to be set via config. 
Additionally:
 - clean api to set remote bitrate estimator mode.
 - clean api to set over use detector options.

R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1448006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4027 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 12:10:58 +00:00
andrew@webrtc.org
d72262dc01 Fix compile errors in ViE with latest clang. 
Rolling to the latest Chromium picks up a new clang, which catches a fresh error:

error: 'reinterpret_cast' to class 'webrtc::VideoEngineImpl *' from its base at non-zero offset 'webrtc::VideoEngine *' behaves differently from 'static_cast' [-Werror,-Wreinterpret-base-class]
 VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
                              ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
../../webrtc/video_engine/vie_codec_impl.cc:36:31: note: use 'static_cast' to adjust the pointer correctly while downcasting
  VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
                              ^~~~~~~~~~~~~~~~
                              static_cast

This was triggered by André's change here:
https://code.google.com/p/webrtc/source/detail?r=3986
which made VideoEngineImpl a derived class of VideoEngine (good).

Picked up one other error as well:
error: implicit conversion from 'long' to 'int' changes value from 9223372036854775807 to -1 [-Werror,-Wconstant-conversion]
        AutoTestSleep(std::numeric_limits<long>::max());
        ~~~~~~~~~~~~~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

This fixes the errors and is required before stable can be rolled in Chromium.

TBR=mflodman,andresp

Review URL: https://webrtc-codereview.appspot.com/1450004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3989 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-09 02:12:07 +00:00
stefan@webrtc.org
8ca8a71de2 Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." 
This reverts commit aae26db1da5803482b094357c546b8454ab1c26d.

BUG=1613
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1327008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3890 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 16:48:32 +00:00
stefan@webrtc.org
ccd4b2aec8 Add a default RTT to CallStats and use different values for buffered/real-time mode. 
BUG=1613

Review URL: https://webrtc-codereview.appspot.com/1326007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3888 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 15:58:23 +00:00
mflodman@webrtc.org
9f5ebb5251 Adding a payload type for RTX. 
BUG=736
TEST=Modified RTP unittests.

Review URL: https://webrtc-codereview.appspot.com/1278004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3843 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 14:55:46 +00:00
pbos@webrtc.org
b238d1210b WebRtc_Word32 -> int32_t in video_engine/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1302005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3801 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:41:51 +00:00
mikhal@webrtc.org
ef9f76a59d Adding a receive side API for buffering mode. 
At the same time, renaming the send side API.

Review URL: https://webrtc-codereview.appspot.com/1104004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3525 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-15 23:22:18 +00:00
mikhal@webrtc.org
3d305c64b4 Updates to send side streaming mode: 
1. Disabling frame-droppers from the vie encoder and not the channel.
2. Accounting for qpMax in the VP8 wrapper.

Review URL: https://webrtc-codereview.appspot.com/1101007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3492 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-10 18:42:55 +00:00
mflodman@webrtc.org
4fd5527ab1 Don't report an error for GetEstimatedReceiveBandwidth if there is no valid 
estimate.

BUG=1377

Review URL: https://webrtc-codereview.appspot.com/1095005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3479 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-06 17:46:39 +00:00
mikhal@webrtc.org
dbe97d2550 Adding a send side API for streaming 
Review URL: https://webrtc-codereview.appspot.com/1070009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3457 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-01 19:33:21 +00:00
mflodman@webrtc.org
d6ec386ff5 Revert the revert in r2988 since that wasn't the issue. 
Review URL: https://webrtc-codereview.appspot.com/931005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2992 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-25 11:30:29 +00:00
vikasmarwaha@webrtc.org
8239ca5096 Reverse Merged r2884 & r2888 from trunk. 
Review URL: https://webrtc-codereview.appspot.com/929005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2988 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-24 22:35:52 +00:00
andrew@webrtc.org
14b43beb7c Move src/ -> webrtc/ 
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-22 18:19:23 +00:00