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25 Commits

Author SHA1 Message Date
Henrik Kjellander
0b9e29c87d Remove include dirs from modules/{media_file,pacing} 
Also move files out of media_file/source.

BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=asapersson@webrtc.org, perkj@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1435093002 .

Cr-Commit-Position: refs/heads/master@{#10647}
 2015-11-16 10:12:32 +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
mflodman
0dbf0090a9 Remove the video channel id completely. 
BUG=webrtc:5079

Review URL: https://codereview.webrtc.org/1412143002

Cr-Commit-Position: refs/heads/master@{#10324}
 2015-10-19 15:12:19 +00:00
Peter Boström
7083e119e8 Remove callback_cs_ in ViEEncoder. 
Instead make callbacks const and set on construction.

BUG=webrtc:1695
R=philipel@webrtc.org

Review URL: https://codereview.webrtc.org/1354143004 .

Cr-Commit-Position: refs/heads/master@{#10017}
 2015-09-22 14:29:00 +00:00
Peter Boström
8e4e8b0455 Simplify BitrateAllocator::AddBitrateObserver. 
Remove start_bitrate_bps which is no longer used and return the current
allocated bitrate instead of having it as an out parameter, removing the
previous return value which is no longer used.

Permits removing bitrate controller usage from ViEEncoder.

BUG=webrtc:1695
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1343783006 .

Cr-Commit-Position: refs/heads/master@{#9942}
 2015-09-15 13:08:12 +00:00
Peter Boström
a753177f93 Remove default ViEEncoder encoder instance. 
BUG=webrtc:1695
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1341943003 .

Cr-Commit-Position: refs/heads/master@{#9940}
 2015-09-15 11:12:32 +00:00
pbos
ef35f069e7 Remove webrtc::Config from ViEChannelGroup. 
Also removing webrtc/experiments.h which is no longer used.

BUG=webrtc:1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1250513006

Cr-Commit-Position: refs/heads/master@{#9642}
 2015-07-27 15:37:14 +00:00
pbos
8ff04d6b3b Remove UpdateSsrcs from EncoderStateFeedback. 
Removes ability to modify set SSRCs from EncoderStateFeedback after
construction.

BUG=webrtc:1695
R=sprang@webrtc.org
TBR=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1241123002

Cr-Commit-Position: refs/heads/master@{#9603}
 2015-07-20 15:01:25 +00:00
Peter Boström
5a3ebd761c Revert "Remove default encoder/decoders." 
This reverts commit 78ae00eea29850bd3bd608287d8b057c88e91b42 due to perf
regressions. Reverting during investigation to figure out root causes.

BUG=chromium:491112
R=henrik.lundin@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/56449004

Cr-Commit-Position: refs/heads/master@{#9283}
 2015-05-26 09:44:14 +00:00
Peter Boström
78ae00eea2 Remove default encoder/decoders. 
This path is not used, senders/receivers already disable default coders.

BUG=1695
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54449004

Cr-Commit-Position: refs/heads/master@{#9245}
 2015-05-21 07:56:17 +00:00
Stefan Holmer
e590416722 Moving the pacer and the pacer thread to ChannelGroup. 
This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out.

BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45549004

Cr-Commit-Position: refs/heads/master@{#8864}
 2015-03-26 10:11:22 +00:00
stefan@webrtc.org
792f1a14e2 Break out allocation from BitrateController into a BitrateAllocator. 
This also refactors some of the padding and allocation code in ViEEncoder, and
makes ChannelGroup a simple forwarder from BitrateController to
BitrateAllocator.

This CL is part of a bigger picture, see https://review.webrtc.org/35319004/ for
details.

BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org, sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44399004

Cr-Commit-Position: refs/heads/master@{#8595}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8595 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-04 12:25:17 +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
9dd0ebc379 Remove the default RTP module. 
This CL removes the default module owned by ViEEncoder, functionality in
the module to register default modules and the final changes in
rtp_rtcp_impl using default/child modules.

BUG=769
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42509004

Cr-Commit-Position: refs/heads/master@{#8514}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8514 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-26 12:58:24 +00:00
pbos@webrtc.org
40367f984b Remove default video encoders for new video API. 
Reduces stream creation time significantly. As a side effect also
removes default encoders for receive-only channels.

BUG=1788,1667
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37049004

Cr-Commit-Position: refs/heads/master@{#8356}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8356 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 08:00:42 +00:00
wuchengli@chromium.org
637c55f45b Add support of texture frames for video capturer. 
This is a reland of r6252. The video_capture_tests failure on
builder Android Chromium-APK Tests should be flaky.

- Add ViECapturer unittest.
- Add CloneFrame function in I420VideoFrame.
- Encoders do not support texture yet and texture frames
are dropped in ViEEncoder for now.

Corresponding CLs:
https://codereview.chromium.org/277943002
http://cl/66620352

BUG=chromium:362437
TEST=WebRTC video stream forwarding, video_engine_core_unittests,
     common_video_unittests and video_capture_tests_apk.
TBR=fischman@webrtc.org, perkj@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6258 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 07:00:51 +00:00
wuchengli@chromium.org
89e8ffb395 Revert "Add support of texture frames for video capturer." 
This reverts commit 83c89cd003be75d7d06ef9a2b139588f08d280ca.

Reason: The Buildbot has detected a new failure on builder
Android Chromium-APK Tests.

BUG=chromium:362437
TBR=fischman@webrtc.org, perkj@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6253 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-27 14:12:58 +00:00
wuchengli@chromium.org
efe15355ee Add support of texture frames for video capturer. 
- Add ViECapturer unittest.
- Add CloneFrame function in I420VideoFrame.
- Encoders do not support texture yet and texture frames
  are dropped in ViEEncoder for now.

Corresponding CLs:
https://codereview.chromium.org/277943002
http://cl/66620352

BUG=chromium:362437
TEST=WebRTC video stream forwarding. Run video_engine_core_unittests and common_video_unittests.
R=fischman@webrtc.org, perkj@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6252 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-27 12:40:27 +00:00
henrike@webrtc.org
79cf3acc79 Removes usage of ListWrapper from several files. 
BUG=2164
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 15:21:30 +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
andresp@webrtc.org
7707d060bb Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1450008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4007 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-13 10:50:50 +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
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