zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
10,455 Commits 1 Branch 0 Tags
3e1cfa7edba8081faada275683b3d1fc71f37ac7
Commit Graph

10455 Commits

This Branch
This Branch
All Branches
Author SHA1 Message Date
nisse
3e1cfa7edb Delete unused method webrtc::VideoRendererInterface::SetSize. 
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1582493002

Cr-Commit-Position: refs/heads/master@{#11223}
 2016-01-12 14:39:25 +00:00
Henrik Boström
3235a27e7a Updated chromium/.gclient and sync_chromium.py to not ignore third_party/ffmpeg. 
Was forgotten in this CL: https://codereview.webrtc.org/1575913003/

BUG=468365
R=kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1579153003 .

Cr-Commit-Position: refs/heads/master@{#11222}
 2016-01-12 14:05:39 +00:00
terelius
2845a02339 Remove unused enum RTPDirections. 
BUG=

Review URL: https://codereview.webrtc.org/1582523002

Cr-Commit-Position: refs/heads/master@{#11221}
 2016-01-12 13:01:02 +00:00
Stefan Holmer
3842c5c7f7 Wire-up BWE feedback for audio receive streams. 
Also wires up receiving transport sequence numbers.

BUG=webrtc:5263
R=mflodman@webrtc.org, pbos@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1535963002 .

Cr-Commit-Position: refs/heads/master@{#11220}
 2016-01-12 12:55:11 +00:00
Peter Boström
6183de6da0 Remove tools/refactoring. 
No longer used, references old GIPS types variable names and confuses
team members which think this code could be used/still useful.

BUG=
R=phoglund@webrtc.org

Review URL: https://codereview.webrtc.org/1581573003 .

Cr-Commit-Position: refs/heads/master@{#11219}
 2016-01-12 12:41:11 +00:00
nisse
127782bbb1 Add default dummy implementation of cricket::VideoRenderer::SetSize, to easy later removal. 
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1581583002

Cr-Commit-Position: refs/heads/master@{#11218}
 2016-01-12 11:39:20 +00:00
Henrik Kjellander
16979e35b3 Update .gitignore 
This should have been done in https://codereview.webrtc.org/1503883002

TBR=phoglund@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/1575113003 .

Cr-Commit-Position: refs/heads/master@{#11217}
 2016-01-12 07:17:49 +00:00
jackychen
67e94fb6f2 Add unit test for stand-alone denoiser and fixed some bugs. 
The unit test will run the pure C denoiser and SSE2/NEON denoiser (based
on the CPU detection) and compare the denoised frames to ensure the bit
exact.

TBR=tommi@webrtc.org

BUG=webrtc:5255

Review URL: https://codereview.webrtc.org/1492053003

Cr-Commit-Position: refs/heads/master@{#11216}
 2016-01-12 05:34:14 +00:00
aluebs
b2328d11dc Remove additional channel constraints when Beamforming is enabled in AudioProcessing 
The general constraints on number of channels for AudioProcessing is:
num_in_channels == num_out_channels || num_out_channels == 1

When Beamforming is enabled and additional constraint was added forcing:
num_out_channels == 1

This artificial constraint was removed by adding upmixing support in CopyTo, since it was already supported for the AudioFrame interface using InterleaveTo.

Review URL: https://codereview.webrtc.org/1571013002

Cr-Commit-Position: refs/heads/master@{#11215}
 2016-01-12 04:32:32 +00:00
kjellander
e93ad1b129 Roll chromium_revision 8c958e0..131167b (368561:368784) 
Change log: 8c958e0..131167b
Full diff: 8c958e0..131167b

Changed dependencies:
* src/third_party/ffmpeg: 58b10df..a41fa51
* src/third_party/libvpx_new/source/libvpx: a9dd8a7..b520882
DEPS diff: 8c958e0..131167b/DEPS

No update to Clang.

TBR=marpan@webrtc.org, stefan@webrtc.org,

Review URL: https://codereview.webrtc.org/1575283002

Cr-Commit-Position: refs/heads/master@{#11214}
 2016-01-12 04:04:35 +00:00
aluebs
2a34688f86 Make Beamforming dynamically settable for Android platform builds 
Review URL: https://codereview.webrtc.org/1563493005

Cr-Commit-Position: refs/heads/master@{#11213}
 2016-01-12 02:04:33 +00:00
andrew
2bc63a1dd3 clang-format audio_device/mac. 
NOTRY=true

Review URL: https://codereview.webrtc.org/1570063003

Cr-Commit-Position: refs/heads/master@{#11212}
 2016-01-11 23:59:25 +00:00
Guo-wei Shieh
a7446d2a50 Change DTLS default from 1.0 to 1.2 for webrtc. 
This changes for standalone webrtc applications.

BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1548733002 .

Cr-Commit-Position: refs/heads/master@{#11211}
 2016-01-11 23:27:12 +00:00
Jon Hjelle
f6c318ebae Update API for Objective-C RTCMediaSource. 
BUG=
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1538263002 .

Patch from Jon Hjelle <hjon@andyet.net>.

Cr-Commit-Position: refs/heads/master@{#11210}
 2016-01-11 22:39:05 +00:00
Jon Hjelle
e799badacc Move Objective-C video renderers to webrtc/api/objc. 
BUG=
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1542473003 .

Patch from Jon Hjelle <hjon@andyet.net>.

Cr-Commit-Position: refs/heads/master@{#11209}
 2016-01-11 21:47:17 +00:00
Jon Hjelle
81028796bc Update API for Objective-C RTCMediaStreamTrack. 
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1527143002 .

Patch from Jon Hjelle <hjon@andyet.net>.

Cr-Commit-Position: refs/heads/master@{#11208}
 2016-01-11 21:16:19 +00:00
Jon Hjelle
a2c353f815 Update API for Objective-C RTCStats. 
BUG=
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1540113002 .

Patch from Jon Hjelle <hjon@andyet.net>.

Cr-Commit-Position: refs/heads/master@{#11207}
 2016-01-11 21:11:45 +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
lally
27ed3cc28c SCTP: Stopped accepting SSRCs higher than max. 
Seems to fix asan-related crash.

BUG=https://code.google.com/p/chromium/issues/detail?id=570261

Review URL: https://codereview.webrtc.org/1571853002

Cr-Commit-Position: refs/heads/master@{#11205}
 2016-01-11 18:24:35 +00:00
hbos
a9a1d2acaf H.264: Default flags and pulling in openh264 and ffmpeg. 
Defining use_third_party_h264 directly, and indirectly defining use_openh264 (from third_party/openh264) and ffmpeg_branding (from third_party/ffmpeg).
These will be used in a follow-up CL that adds an encoder and decoder implementation.
The flags are added in this CL so that they can be used by trybots/waterfall bots in GN without "Build argument had no effect" errors. Equivalent GYP changes are also added.

BUG=468365

Review URL: https://codereview.webrtc.org/1575913003

Cr-Commit-Position: refs/heads/master@{#11204}
 2016-01-11 18:19:06 +00:00
Jon Hjelle
7823495698 Move RTCI420Frame to webrtc/api/objc/RTCVideoFrame with minor style changes. 
BUG=
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1533193003 .

Patch from Jon Hjelle <hjon@andyet.net>.

Cr-Commit-Position: refs/heads/master@{#11203}
 2016-01-11 17:47:14 +00:00
kjellander
fd99dea4f6 Roll chromium_revision 42ab10e..8c958e0 (368534:368561) 
Change log: 42ab10e..8c958e0
Full diff: 42ab10e..8c958e0

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1575003002

Cr-Commit-Position: refs/heads/master@{#11202}
 2016-01-11 12:57:26 +00:00
danilchap
ef3d805f6e [rtp_rtcp] rtcp::Tmmbn moved into own file 
explicetly unchanged.

BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1578713002

Cr-Commit-Position: refs/heads/master@{#11201}
 2016-01-11 11:31:17 +00:00
kjellander
d36efeb622 Roll chromium_revision e738b54..42ab10e (368533:368534) 
Change log: e738b54..42ab10e
Full diff: e738b54..42ab10e

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1573883002

Cr-Commit-Position: refs/heads/master@{#11200}
 2016-01-11 03:56:01 +00:00
kjellander
4de003722d Roll chromium_revision 7d97c94..e738b54 (368514:368533) 
Change log: 7d97c94..e738b54
Full diff: 7d97c94..e738b54

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1572173002

Cr-Commit-Position: refs/heads/master@{#11199}
 2016-01-10 11:58:43 +00:00
kjellander
3c05e6c9c3 Disable EndToEndTest.TransportSeqNumOnAudioAndVideo for Dr Memory. 
It started failing at the roll in
https://codereview.webrtc.org/1556273002

BUG=webrtc:5402
TBR=marpan@webrtc.org
NOTRY=True

Review URL: https://codereview.webrtc.org/1574813002

Cr-Commit-Position: refs/heads/master@{#11198}
 2016-01-10 05:46:42 +00:00
guoweis
daa87497e8 Revert of Roll chromium_revision 7d97c94..951c006 (368514:368525) (patchset #1 id:1 of https://codereview.webrtc.org/1577573002/ ) 
Reason for revert:
Win DrMemory Full: video_engine_tests failed 1

https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/builds/3465

Original issue's description:
> Roll chromium_revision 7d97c94..951c006 (368514:368525)
>
> Change log: 7d97c94..951c006
> Full diff: 7d97c94..951c006
>
> No dependencies changed.
> No update to Clang.
>
> TBR=
>
> Committed: https://crrev.com/6109fc13aadebf7c5a990bbc78e981ab215321a6
> Cr-Commit-Position: refs/heads/master@{#11195}

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1570403002

Cr-Commit-Position: refs/heads/master@{#11197}
 2016-01-10 03:27:35 +00:00
Guo-wei Shieh
db21f633a2 fix GN build break on native_client 
TBR=guidou@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/1576723002 .

Cr-Commit-Position: refs/heads/master@{#11196}
 2016-01-09 21:12:11 +00:00
kjellander
6109fc13aa Roll chromium_revision 7d97c94..951c006 (368514:368525) 
Change log: 7d97c94..951c006
Full diff: 7d97c94..951c006

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1577573002

Cr-Commit-Position: refs/heads/master@{#11195}
 2016-01-09 19:59:36 +00:00
kjellander
0697db6f1b Roll chromium_revision 8a15a7f..7d97c94 (368391:368514) 
Change log: 8a15a7f..7d97c94
Full diff: 8a15a7f..7d97c94

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1577543002

Cr-Commit-Position: refs/heads/master@{#11194}
 2016-01-09 04:09:33 +00:00
Guo-wei Shieh
684e995464 Disable 2 video tests which fail on DrMemoryFull 
TBR=kjellander@webrtc.org
BUG=5417

Review URL: https://codereview.webrtc.org/1575433003 .

Cr-Commit-Position: refs/heads/master@{#11193}
 2016-01-09 03:03:50 +00:00
Taylor Brandstetter
f475d365a2 Properly handle different transports having different SSL roles. 
This meant splitting "transport_options" into audio/video/data options,
for when creating the answer, and giving "GetSslRole" a "transport_name"
parameter so we can retrieve the current role on a per-transport basis.

BUG=webrtc:4525
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1516993002 .

Cr-Commit-Position: refs/heads/master@{#11192}
 2016-01-08 23:36:06 +00:00
pkasting
25702cb162 Misc. small cleanups. 
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests).  Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks

All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.

BUG=none
TEST=none

Review URL: https://codereview.webrtc.org/1534193008

Cr-Commit-Position: refs/heads/master@{#11191}
 2016-01-08 21:50:32 +00:00
kjellander
5de688ed34 Roll chromium_revision ede5d4f..8a15a7f (368310:368391) 
Change log: ede5d4f..8a15a7f
Full diff: ede5d4f..8a15a7f

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1573583002

Cr-Commit-Position: refs/heads/master@{#11190}
 2016-01-08 21:09:36 +00:00
minyue
49c454e748 Cleaning neteq_unittest resource files. 
BUG=webrtc:2692

Review URL: https://codereview.webrtc.org/1563983003

Cr-Commit-Position: refs/heads/master@{#11189}
 2016-01-08 19:30:18 +00:00
kjellander
f1685c771d Disable RampUpTest.UpDownUp* in webrtc_perf_tests on Mac 
NOTRY=True
BUG=5407
TBR=stefan@webrtc.org,pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1569273003

Cr-Commit-Position: refs/heads/master@{#11188}
 2016-01-08 18:43:45 +00:00
stefan
e74eef19bd Add CreateSend/ReceiveTransport() methods to CallTest. 
This allows the test to create its own transports if it, for instance, needs to do demuxing.

BUG=webrtc:5416

Review URL: https://codereview.webrtc.org/1573453002

Cr-Commit-Position: refs/heads/master@{#11187}
 2016-01-08 14:47:21 +00:00
phoglund
37ebcf0ce5 Reland "Add APK targets to build libjingle tests for Android." 
patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/

This reverts commit bc14164aad254e72ce4d1e381b912b7d3acf5391.

We have made more preparations downstream, so this should work now. Original CL by perkj@.

BUG=webrtc:2365
The work started from the work by kjellander@ in https://codereview.webrtc.org/1413663003/

Review URL: https://codereview.webrtc.org/1570513004

Cr-Commit-Position: refs/heads/master@{#11186}
 2016-01-08 13:05:01 +00:00
kjellander
b71b4f0c7a Update attributes to match gclibc's ansidecl.h 
To ease use of WebRTC in other codebases, update some macros
to match glibc's ansidecl.h, which uses double-underscores for attributes.

NOTRY=True

Review URL: https://codereview.webrtc.org/1571653002

Cr-Commit-Position: refs/heads/master@{#11185}
 2016-01-08 12:51:47 +00:00
kjellander
004851caca Roll chromium_revision 32569c6..ede5d4f (368258:368310) 
Change log: 32569c6..ede5d4f
Full diff: 32569c6..ede5d4f

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1567393002

Cr-Commit-Position: refs/heads/master@{#11184}
 2016-01-08 11:59:15 +00:00
henrik.lundin
e1ca167217 Add tracing to NetEqImpl::GetAudio 
BUG=webrtc:5167
R=pbos@webrtc.org
NOTRY=true

Review URL: https://codereview.webrtc.org/1571693002

Cr-Commit-Position: refs/heads/master@{#11183}
 2016-01-08 11:50:14 +00:00
andrew
ec80f03b3c Check the mic volume only periodically on Mac. 
Ask the OS for the mic volume every 1 second rather than with every 10
ms chunk. The previous behavior was consuming ~2% of the CPU load of
a voice engine call, and is now negligible.

This is consistent with the webrtc Windows Core Audio implementation,
as well as the Chromium Mac implementation:
https://code.google.com/p/chromium/codesearch#chromium/src/media/audio/agc_audio_stream.h

TEST=voe_cmd_test with AGC continues to work well on Mac.

Review URL: https://codereview.webrtc.org/1564223002

Cr-Commit-Position: refs/heads/master@{#11182}
 2016-01-08 09:16:25 +00:00
perkj
fbeb97e01f Fix clang warning in peerconnection_jni.cc 
TEST= export GYP_DEFINES="OS=android clang=1" ...
      ninja -C out/Debug AppRTCDemo
BUG=webrtc:5399

Review URL: https://codereview.webrtc.org/1561073005

Cr-Commit-Position: refs/heads/master@{#11181}
 2016-01-08 08:43:15 +00:00
asapersson
59bac1a4c5 Fix for stats updated twice when switching content type (realtime <-> screenshare). Add unittest. 
BUG=

Review URL: https://codereview.webrtc.org/1543933004

Cr-Commit-Position: refs/heads/master@{#11180}
 2016-01-08 07:36:06 +00:00
kjellander
95ab30ce88 Roll chromium_revision 6dd04c2..32569c6 (368115:368258) 
Change log: 6dd04c2..32569c6
Full diff: 6dd04c2..32569c6

Changed dependencies:
* src/third_party/libyuv: 1ccbf8f..fc52d8d
DEPS diff: 6dd04c2..32569c6/DEPS

No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1570713002

Cr-Commit-Position: refs/heads/master@{#11179}
 2016-01-08 07:31:13 +00:00
kjellander
a2b1e03c66 Disable AudioDeviceAPITest.MicrophoneVolumeTests on Linux. 
NOTRY=True
BUG=webrtc:5414
TBR=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1572503002

Cr-Commit-Position: refs/heads/master@{#11178}
 2016-01-08 07:26:11 +00:00
Taylor Brandstetter
893505d0fb Adding unit test to ensure TURN server priorities are unique. 
BUG=webrtc:5209
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1570563002 .

Cr-Commit-Position: refs/heads/master@{#11177}
 2016-01-07 23:12:53 +00:00
Taylor Brandstetter
e5ba13bc09 Adding a way for a Java RtpSender to set a track without taking ownership. 
This means that the track will still have a reference count after the
PeerConnection and RtpSender have been destroyed.

R=glaznev@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1566103003 .

Cr-Commit-Position: refs/heads/master@{#11176}
 2016-01-07 23:11:33 +00:00
kjellander
ced8ec9069 Roll chromium_revision bd5949f..6dd04c2 (368055:368115) 
Change log: bd5949f..6dd04c2
Full diff: bd5949f..6dd04c2

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1568963002

Cr-Commit-Position: refs/heads/master@{#11175}
 2016-01-07 21:22:52 +00:00
A.Brauckmann
bedc17be5c Fixing integer underflow in FileAudioDevice (webrtc issue 4554) 
Problem is described here:
https://code.google.com/p/webrtc/issues/detail?id=4554

Review URL: https://codereview.webrtc.org/1295603002

Cr-Commit-Position: refs/heads/master@{#11174}
 2016-01-07 20:38:36 +00:00