23 Commits

Author SHA1 Message Date
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
danilchap
7c9426cf38 Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
Review URL: https://codereview.webrtc.org/1877253002

Cr-Commit-Position: refs/heads/master@{#12359}
2016-04-14 10:05:37 +00:00
solenberg
1d0313916b Reland https://codereview.webrtc.org/1802993002/
Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.

BUG=webrtc:4690

Committed: https://crrev.com/69a81999ace08e40e2b2ec526b0e111aa11b9538
Cr-Commit-Position: refs/heads/master@{#12015}

Review URL: https://codereview.webrtc.org/1840893004

Cr-Commit-Position: refs/heads/master@{#12157}
2016-03-30 09:42:37 +00:00
solenberg
b69395b374 Revert of Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code. (patchset #2 id:20001 of https://codereview.webrtc.org/1802993002/ )
Reason for revert:
Revert because it breaks downstream code.

Original issue's description:
> Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/69a81999ace08e40e2b2ec526b0e111aa11b9538
> Cr-Commit-Position: refs/heads/master@{#12015}

TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1812453002

Cr-Commit-Position: refs/heads/master@{#12016}
2016-03-16 14:05:21 +00:00
solenberg
69a81999ac Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1802993002

Cr-Commit-Position: refs/heads/master@{#12015}
2016-03-16 12:59:04 +00:00
Peter Kasting
6955870806 Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1316523002 .

Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-13 00:26:55 +00:00
danilchap
6db6cdc604 [rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
BUG=webrtc:5277
R=mflodman

Review URL: https://codereview.webrtc.org/1513303003

Cr-Commit-Position: refs/heads/master@{#11025}
2015-12-15 10:54:50 +00:00
Henrik Kjellander
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
Henrik Kjellander
98f53510b2 system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1413333002 .

Cr-Commit-Position: refs/heads/master@{#10438}
2015-10-28 17:17:50 +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
pbos
d436298332 Remove ResetStatistics from RTP feedback.
BUG=
R=asapersson@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1213603002

Cr-Commit-Position: refs/heads/master@{#9548}
2015-07-07 15:32:56 +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
pkasting@chromium.org
4591fbd09f Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
stefan@webrtc.org
6071b0636d Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such.
This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.

This also highlighted a number of unused functions which I've removed.

-- This is was reviewed in https://webrtc-codereview.appspot.com/19309004/, but
-- a new cl was needed to resolve a small conflict before committing.

BUG=none
TEST=none
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7162 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-12 07:42:33 +00:00
henrik.lundin@webrtc.org
1972ff8a6e Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.
This will make a subsequent change I intend to do safer, where I'll change the
return type of one of the base Module functions, by breaking the compile if I
miss any overrides.

This also highlighted a number of unused functions (in many cases apparently
virtual "overrides" of no-longer-existent base functions).  I've removed some of
these.

This also highlighted several cases where "virtual" was used unnecessarily to
mark a function that was only defined in one class.  Removed "virtual" in those
cases.

BUG=none
TEST=none
R=andrew@webrtc.org, henrik.lundin@webrtc.org, mallinath@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7146 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 06:20:28 +00:00
stefan@webrtc.org
48df38114d Fix for making sure that the packet in order checks are done prior to updating the last received packet state.
Without this fix all packets are considered out-of-order by the rtp receiver, causing the last received state
in the rtp receiver to never get valid.

Also makes sure that only valid timestamps and receive times are used for audio/video sync.

BUG=2608
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5102 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-08 15:18:52 +00:00
stefan@webrtc.org
7bb8f02274 Adds support for combining RTX and FEC/RED.
This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX.

Enables retransmissions over RTX by default in the loopback test.

BUG=1811
TESTS=voe/vie_auto_test --automated and trybots.
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2154004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-06 13:40:11 +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