Commit Graph

28 Commits

Author SHA1 Message Date
a8890a57a5 rtcp::Nack packet moved into own file and got Parse function
Review URL: https://codereview.webrtc.org/1461623003

Cr-Commit-Position: refs/heads/master@{#11111}
2015-12-22 11:43:10 +00:00
54999d411b rtcp::Dlrr block moved into own file and got Parse function
BUG=webrtc:5260

Review URL: https://codereview.webrtc.org/1453973005

Cr-Commit-Position: refs/heads/master@{#11044}
2015-12-16 09:56:22 +00:00
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
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
162abd3562 lint whitespace warning removed from most rtp_rtcp/source/ files
rtcp_utility, rtp_utility, tmmbr_help, rtcp_receiver, rtcp_receiver_help are explicetly excluded from the cleanup becaues there are short plans (or cls) to do a deeper cleaning there.

BUG=webrtc:5277
R=pbos@webrtc.org, mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1512493002

Cr-Commit-Position: refs/heads/master@{#10966}
2015-12-10 10:39:45 +00:00
5eb4988c0a [rtp_rtcp] Lint build/header_guard errors fixed
BUG=webrtc:5277
R=mflodman

Review URL: https://codereview.webrtc.org/1506043003

Cr-Commit-Position: refs/heads/master@{#10949}
2015-12-09 11:32:45 +00:00
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
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
f7c5776d42 Refactorings to send RTCP packets directly via the RtcpPacket callback, with some simplifications enabled by this. NACK now also sent via RtcpPacket.
BUG=webrtc:2450
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1309833002 .

Cr-Commit-Position: refs/heads/master@{#10888}
2015-12-04 09:40:54 +00:00
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
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
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
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
df948f03b3 rtcp::ReportBlock refactored to contain parsing
Review URL: https://codereview.webrtc.org/1420283022

Cr-Commit-Position: refs/heads/master@{#10633}
2015-11-13 11:03:18 +00:00
ff761fba82 modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

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

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00
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
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
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
3c089d751e Add RTC_ prefix to contructormagic macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.

* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS

Related CL: https://codereview.webrtc.org/1335923002/

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1345433002

Cr-Commit-Position: refs/heads/master@{#9953}
2015-09-16 12:37:52 +00:00
73a93e8257 Add a ParseHeader method to RtcpPacket, for parsing common RTCP header.
Also refactor TransportFeedback to use this.

BUG=

Review URL: https://codereview.webrtc.org/1307663004

Cr-Commit-Position: refs/heads/master@{#9935}
2015-09-14 19:50:49 +00:00
a3b8769860 Add packetization and coding/decoding of feedback message format.
BUG=webrtc:4312
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1175263002 .

Cr-Commit-Position: refs/heads/master@{#9651}
2015-07-29 08:47:04 +00:00
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
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
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
3b84b3a58c Add RTCP packet types to packet builder:
REMB, TMMBR, TMMBN and
extended reports: RRTR, DLRR, VoIP metric.

BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9299005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6537 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-25 12:22:17 +00:00
4b12d40008 Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class.
BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6449 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 14:09:28 +00:00
a826006132 Add NACK and RPSI packet types to RTCP packet builder.
Fixes bug found when parsing received RPSI packet.

BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6194 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 09:53:51 +00:00
0f2809a5ac Add RTCP packet class.
Adds packet types: sr, rr, bye, fir.

BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5592 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-21 08:14:45 +00:00