77 Commits

Author SHA1 Message Date
Erik Språng
737336d37a Add NACK rate throttling for audio channels.
Not really used for audio today (already in place for video), but should
still function anyway.

BUG=
R=henrika@webrtc.org, minyue@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2181383002 .

Cr-Commit-Position: refs/heads/master@{#13571}
2016-07-29 10:59:49 +00:00
kwiberg
84be511ac0 Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/
(This is a re-land of https://codereview.webrtc.org/1921233002, which
got reverted for breaking Chromium.)

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1923133002

Cr-Commit-Position: refs/heads/master@{#12522}
2016-04-27 08:20:08 +00:00
terelius
52d4e6bf5e Revert of Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/ (patchset #1 id:40001 of https://codereview.webrtc.org/1921233002/ )
Reason for revert:
Fails on Chromium FYI bots.

https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/5392/

Original issue's description:
> Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/
>
> BUG=webrtc:5520
>
> Committed: https://crrev.com/2c27a062ee46258abe9facc2cceee74f09bf6a99
> Cr-Commit-Position: refs/heads/master@{#12511}

TBR=tommi@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1924443002

Cr-Commit-Position: refs/heads/master@{#12513}
2016-04-26 16:32:09 +00:00
kwiberg
2c27a062ee Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1921233002

Cr-Commit-Position: refs/heads/master@{#12511}
2016-04-26 15:38:03 +00:00
Peter Boström
d98d457134 Remove "This file includes unit tests" comments.
Superfluous, neteq_unittests.cc conveys that the file contains unittests
for NetEq for instance.

BUG=
TBR=henrik.lundin@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1922103002 .

Cr-Commit-Position: refs/heads/master@{#12504}
2016-04-26 11:44:02 +00:00
Danil Chapovalov
ee6e4272a4 Fixed undefined shift in parsing Tmmbr, Tmmbn and Remb
BUG=chromium:603483
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1888793003 .

Cr-Commit-Position: refs/heads/master@{#12423}
2016-04-19 10:15:21 +00:00
danilchap
d6b851a1bd Fixed memleak when two voip blocks present in single rtcp packet.
BUG=chromium:603894

Review URL: https://codereview.webrtc.org/1901593002

Cr-Commit-Position: refs/heads/master@{#12413}
2016-04-18 17:54:13 +00:00
danilchap
90a1351072 Fixed rtcp rpsi parsing of invalid packets.
Added packet type RpsiItem to destinguish parsed rpsi header and Rpsi body
  preventing handling two half-valid (header-only) rpsi packets as one valid,
  making test parser calculate rpsi packet once instead of twice.
Added check padding bits doesn't exceed native bit string length.
Marking rpsi received moved after it is validated.

BUG=600977

Review URL: https://codereview.webrtc.org/1880443002

Cr-Commit-Position: refs/heads/master@{#12318}
2016-04-11 17:05:06 +00:00
danilchap
7a4116aa54 [rtp_rtcp] Append functionality moved from base RtcpPacket class to CompoundPacket
BUG=webrtc:5260

Review URL: https://codereview.webrtc.org/1582323005

Cr-Commit-Position: refs/heads/master@{#11981}
2016-03-14 15:19:31 +00:00
danilchap
50da1d329a Fixed busy loop in case of partially malformed rtcp packet
BUG=webrtc:5574

Review URL: https://codereview.webrtc.org/1726573003

Cr-Commit-Position: refs/heads/master@{#11946}
2016-03-10 21:13:57 +00:00
Danil Chapovalov
c1e55c7136 rtt calculation handles time go backwards
CompactNtpIntervalToMs renamed to CompactNtpRttToMs and handle special cases:
large values consider negative/invalid and result in value of 1.
0 result consider too small and increases to 1.

BUG=590996
R=asapersson@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1763823003 .

Cr-Commit-Position: refs/heads/master@{#11928}
2016-03-09 14:14:45 +00:00
Danil Chapovalov
a094fd1550 RTT intermediate calculation use ntp time instead of milliseconds.
Compact NTP representation was designed exactly for that purpose: calculate RTT. No need to map to ms before doing arithmetic on this values.
  Because of this change there is no need to keep mapping between compact ntp presentation and milliseconds in the RTCPSender.

BUG=webrtc:5565
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1491843004 .

Cr-Commit-Position: refs/heads/master@{#11710}
2016-02-22 17:59:47 +00:00
danilchap
69e59e619a [rtp_rtcp] rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer
rtcp::RawPacket is rtc::Buffer, it had no extra functionality.
rtc::Buffer is a movable class - no point to wrap it into rtc::scoped_ptr
change is large, but straightforward:
  rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer
  ->Buffer() replaced with .data()
  ->Length() replaced with .size()

BUG=webrtc:5260

Review URL: https://codereview.webrtc.org/1696203002

Cr-Commit-Position: refs/heads/master@{#11649}
2016-02-17 11:11:50 +00:00
danilchap
f174e3a260 [rtp_rtcp] rtcp::Tmmbr cleaned and got Parse function
Added accessor and Parse function
removed dependencies on structures from rtcp_utility.h (except RtcpCommonHeader)

BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1675583002

Cr-Commit-Position: refs/heads/master@{#11502}
2016-02-05 12:56:40 +00:00
danilchap
de13882d94 rtcp::ExtenededReports packet class got Parse function
BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1557593002

Cr-Commit-Position: refs/heads/master@{#11428}
2016-01-29 19:26:20 +00:00
Danil Chapovalov
32e590ec13 class doesn't rely on structures in RTCPUtility to store data.
supports several fci items in same packet.
got accessors to read data

BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1544403002 .

Cr-Commit-Position: refs/heads/master@{#11354}
2016-01-22 10:05:10 +00:00
danilchap
34ed2b95a5 [rtp_rtcp] rtcp::SenderReport moved into own file and got Parse function
BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1544983002

Cr-Commit-Position: refs/heads/master@{#11288}
2016-01-18 10:43:38 +00:00
Danil Chapovalov
1567d0bd98 [rtp_rtcp] rtcp::Sdes moved into own file
Cleaning/Parsing will be done in the https://codereview.webrtc.org/1439553003/

BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1592763002 .

Cr-Commit-Position: refs/heads/master@{#11274}
2016-01-15 16:34:32 +00:00
Danil Chapovalov
256e5b23f8 Cleaning/Parsing will be done in the https://codereview.webrtc.org/1557593002/
BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1579213005 .

Cr-Commit-Position: refs/heads/master@{#11271}
2016-01-15 13:16:36 +00:00
Danil Chapovalov
5679da1291 [rtp_rtcp] rtcp::Fir moved into own file
Cleaning/Parsing will be done in the https://codereview.webrtc.org/1544403002

BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1581983003 .

Cr-Commit-Position: refs/heads/master@{#11269}
2016-01-15 12:19:59 +00:00
Danil Chapovalov
a5eba6c98b [rtp_rtcp] rtcp::Remb moved into own file
Cleaning/Parsing will be done in the https://codereview.webrtc.org/1552773002/

BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1590883002 .

Cr-Commit-Position: refs/heads/master@{#11268}
2016-01-15 11:40:27 +00:00
danilchap
92e677a1f8 [rtp_rtcp] rtcp::Sli packet moved into own file and got Parse function
BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1551893002

Cr-Commit-Position: refs/heads/master@{#11228}
2016-01-12 18:05:00 +00:00
danilchap
7e8145f05d [rtp_rtcp] rtcp::Tmmbr moved into own file
BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1575023002

Cr-Commit-Position: refs/heads/master@{#11206}
2016-01-11 19:49:24 +00:00
danilchap
91941ae493 rtcp::VoipMetric block moved into own file and got Parse function
Review URL: https://codereview.webrtc.org/1452733002

Cr-Commit-Position: refs/heads/master@{#11030}
2015-12-15 15:06:44 +00:00
danilchap
b8b6fbb7a5 lint build/include errors fixed in rtp_rtcp module
BUG=webrtc:5277
R=mflodman

Review URL: https://codereview.webrtc.org/1505993003

Cr-Commit-Position: refs/heads/master@{#10971}
2015-12-10 13:05:35 +00:00
Danil Chapovalov
fc47ed6c05 rtcp::Rrtr block moved into own file and got Parse function
BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1496883002 .

Cr-Commit-Position: refs/heads/master@{#10912}
2015-12-07 13:46:42 +00:00
Danil Chapovalov
97f7e13c23 rtcp::ReceiverReport moved into own file and got Parse function
BUG=webrtc:5260
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1453083002 .

Cr-Commit-Position: refs/heads/master@{#10897}
2015-12-04 15:13:40 +00:00
danilchap
f8385aded0 rtcp::Pli moved into own file and got a Parse function
Created rtcp::Psfb abstract class between rtcp::Pli and rtcp::RtcpPacket to hold common data for Feedback Message.

BUG=webrtc:5260

Review URL: https://codereview.webrtc.org/1446513002

Cr-Commit-Position: refs/heads/master@{#10823}
2015-11-27 13:36:17 +00:00
stefan
64c0a0a111 Revert of Make overuse estimator one dimensional. (patchset #5 id:80001 of https://codereview.webrtc.org/1376423002/ )
Reason for revert:
Broke webrtc_perf_tests on bots.

Original issue's description:
> Make overuse estimator one dimensional.
>
> This drops the payload size difference dimension of the Kalman filter,
> which doesn't improve the quality of the estimation when pacing packets
> on the send-side.
>
> R=gaetano.carlucci@gmail.com, mflodman@webrtc.org, terelius@webrtc.org
>
> Committed: https://crrev.com/06e05a85b9e4def75ed5e6b582c4df842616f25f
> Cr-Commit-Position: refs/heads/master@{#10809}

TBR=terelius@webrtc.org,mflodman@webrtc.org,gaetano.carlucci@gmail.com
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1481003002

Cr-Commit-Position: refs/heads/master@{#10816}
2015-11-27 09:02:35 +00:00
Stefan Holmer
06e05a85b9 Make overuse estimator one dimensional.
This drops the payload size difference dimension of the Kalman filter,
which doesn't improve the quality of the estimation when pacing packets
on the send-side.

R=gaetano.carlucci@gmail.com, mflodman@webrtc.org, terelius@webrtc.org

Review URL: https://codereview.webrtc.org/1376423002 .

Cr-Commit-Position: refs/heads/master@{#10809}
2015-11-26 14:35:10 +00:00
danilchap
50c5136cb2 RTCP Bye packet moved to own file
Bye class got support for Parsing
 Reason field implemented

Review URL: https://codereview.webrtc.org/1430013003

Cr-Commit-Position: refs/heads/master@{#10741}
2015-11-22 17:03:16 +00:00
danilchap
0219c9b4bf rtcp::App moved into own file and got Parse function
Review URL: https://codereview.webrtc.org/1437353003

Cr-Commit-Position: refs/heads/master@{#10688}
2015-11-18 13:56:57 +00:00
danilchap
f8506cbdd8 rtcp::Ij renamed to rtcp::ExtendedJitterReport
to match name given in the RFC5450
  private member renamed to inter_arrival_jitters_ for the same reason.
rtcp::ExtendedJitterReport moved into own file
accessors and Parse function added
  to make class usable for parsing packet

Review URL: https://codereview.webrtc.org/1434213004

Cr-Commit-Position: refs/heads/master@{#10636}
2015-11-13 15:33:26 +00:00
stefan
1d8a506405 Add a PacketOptions struct to webrtc::Transport.
This allows us to pass packet meta data, such as transport sequence
number, to libjingle and further down to the socket implementation. A
similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h.

BUG=4173

Review URL: https://codereview.webrtc.org/1376673004

Cr-Commit-Position: refs/heads/master@{#10144}
2015-10-02 10:39:40 +00:00
sprang
49f9cdba02 Fix bug where rtcp::TransportFeedback may generate incorrect messages.
In particular, if 14 short deltas were inserted (2 * capacity of status
vector chunk with 2bit items) followed by a large delta, that status
item would be dropped.

BUG=

Review URL: https://codereview.webrtc.org/1367193002

Cr-Commit-Position: refs/heads/master@{#10132}
2015-10-01 10:07:04 +00:00
pbos
2d566686a2 Unify Transport and newapi::Transport interfaces.
BUG=webrtc:1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1369263002

Cr-Commit-Position: refs/heads/master@{#10096}
2015-09-28 16:59:36 +00:00
stefan
4fbd145dce Fix suspend below min bitrate in new API by making it possible to set min bitrate at the receive-side.
In addition to this the ramp-up tests are refactored to use a receive call instead of only a remote bitrate estimator, and to make use of BaseTest.

BUG=webrtc:4836

Review URL: https://codereview.webrtc.org/1368943002

Cr-Commit-Position: refs/heads/master@{#10087}
2015-09-28 10:57:23 +00:00
Erik Språng
6b8d355168 Reland "Wire up send-side bandwidth estimation."
Revert was patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/

The culprit was RTC_DCHECK(poller_thread_->Start()); in rampup_test.cc

BUG=webrtc:4173
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1362303002 .

Cr-Commit-Position: refs/heads/master@{#10052}
2015-09-24 13:07:17 +00:00
Erik Språng
c9bbeb0354 Revert of Wire up send-side bandwidth estimation. (patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/ )
Reason for revert:
Breaking some Android bots.
https://chromegw.corp.google.com/i/client.webrtc/builders/Android32%20Tests%20%28L%20Nexus5%29

Original issue's description:
> Wire up send-side bandwidth estimation.
>
> BUG=webrtc:4173
>
> Committed: https://crrev.com/ef165eefc79cf28bb67779afe303cc2365885547
> Cr-Commit-Position: refs/heads/master@{#10012}

TBR=stefan@webrtc.org, kjellander@webrtc.org
NOPRESUBMIT=false
NOTREECHECKS=false
NOTRY=false
BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1362923002 .

Cr-Commit-Position: refs/heads/master@{#10029}
2015-09-23 11:52:01 +00:00
sprang
ef165eefc7 Wire up send-side bandwidth estimation.
BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1338203003

Cr-Commit-Position: refs/heads/master@{#10012}
2015-09-22 12:10:58 +00:00
Peter Boström
ac547a6538 Remove channel ids from various interfaces.
Starts by removing channel/engine id from ViEChannel which propagates
down to the RTP/RTCP module as well as the transport class.

IncomingVideoStream::RenderFrame() is untouched for now but receives a
fake id instead of the previous channel id. Added a TODO to remove it
later but the RenderFrame call is implemented in a lot of
platform-dependent files and should probably remove the "manager" aspect
of renderers, so preferring to do it separately

BUG=webrtc:1695
R=henrika@webrtc.org, mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1335353005 .

Cr-Commit-Position: refs/heads/master@{#9978}
2015-09-17 21:06:02 +00:00
Erik Språng
468e62a974 Remove MimdRateControl and factories for RemoteBitrateEstimor.
BUG=
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1208083002.

Cr-Commit-Position: refs/heads/master@{#9541}
2015-07-06 08:51:01 +00:00
Erik Språng
bdc0b0d869 Use RtcpPacket classes for SenderReport/ReceiveReport in RTCPSender
BUG=2450
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1170723002.

Cr-Commit-Position: refs/heads/master@{#9483}
2015-06-22 13:21:40 +00:00
Erik Språng
c1b9d4e686 Add support for fragmentation in RtcpPacket.
If the buffer becomes full an OnPacketReady callback will be used to
send the packets created so far. On success the buffer can be reused.
The same callback will be called when the last packet has beed created.

Also made some changes to RawPacket. Buffer will now be heap-allocated
rather than (potentially) stack-allocated, but on the plus side it can
now be allocted with variable size and also avoids one memcpy.

BUG=

patch from issue 56429004 at patchset 160001 (http://crrev.com/56429004#ps160001)

R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1165113002

Cr-Commit-Position: refs/heads/master@{#9390}
2015-06-08 07:54:24 +00:00
Peter Boström
fe7a80c38c Prevent sender RTCP signals for receive-only channels.
Since RTCP packets are delivered to both senders and receivers that
correspond the receivers currently log that NACKed packets are missing,
since they have no direct connection to the sending side or the RTP
packet history. Also preventing triggering on SR requests and PLI/FIR.

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

Review URL: https://webrtc-codereview.appspot.com/45249004

Cr-Commit-Position: refs/heads/master@{#9071}
2015-04-23 15:52:58 +00:00
kjellander@webrtc.org
14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 13:04:54 +00:00
kwiberg@webrtc.org
00b8f6b364 Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36229004

Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 14:43:50 +00:00
mflodman@webrtc.org
96abda0316 Removing FEC functionality from the default RTP module.
This CL removes the last default module methods used from ViEEncoder and
the default module itself will be removed in a separate CL.

BUG=769
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35309004

Cr-Commit-Position: refs/heads/master@{#8505}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8505 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 13:50:51 +00:00
pbos@webrtc.org
1d0fa5d352 Add RtcpPacketTypeCounter stats to new API.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788

Review URL: https://webrtc-codereview.appspot.com/37489004

Cr-Commit-Position: refs/heads/master@{#8429}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8429 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 12:47:45 +00:00
pkasting@chromium.org
16825b1a82 Use int64_t more consistently for times, in particular for RTT values.
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t.  Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 21:51:21 +00:00