braveyao@webrtc.org
ebdfa8dcba
Add fischman into OWNERS of WebRTCDemo Android.
...
BUG=
TBR=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1450005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3991 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-09 07:30:38 +00:00
andrew@webrtc.org
d72262dc01
Fix compile errors in ViE with latest clang.
...
Rolling to the latest Chromium picks up a new clang, which catches a fresh error:
error: 'reinterpret_cast' to class 'webrtc::VideoEngineImpl *' from its base at non-zero offset 'webrtc::VideoEngine *' behaves differently from 'static_cast' [-Werror,-Wreinterpret-base-class]
VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
../../webrtc/video_engine/vie_codec_impl.cc:36:31: note: use 'static_cast' to adjust the pointer correctly while downcasting
VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
^~~~~~~~~~~~~~~~
static_cast
This was triggered by André's change here:
https://code.google.com/p/webrtc/source/detail?r=3986
which made VideoEngineImpl a derived class of VideoEngine (good).
Picked up one other error as well:
error: implicit conversion from 'long' to 'int' changes value from 9223372036854775807 to -1 [-Werror,-Wconstant-conversion]
AutoTestSleep(std::numeric_limits<long>::max());
~~~~~~~~~~~~~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This fixes the errors and is required before stable can be rolled in Chromium.
TBR=mflodman,andresp
Review URL: https://webrtc-codereview.appspot.com/1450004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3989 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-09 02:12:07 +00:00
andrew@webrtc.org
c6a3755ada
Update SincResampler with the latest Chromium code.
...
* Brings in on-the-fly sample ratio updates (or varispeed) with minor modifications to build in webrtc.
* Moved SSE and NEON optimized functions into their own files to handle run-time detection properly. NEON optimizations now enabled.
TESTED=unit tests and ran voe_cmd_test loopback with both devices using 44.1 kHz to exercise SincResampler in real-time.
R=dalecurtis@chromium.org , kma@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1438004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3987 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-08 20:35:43 +00:00
andresp@webrtc.org
44272739c2
Clean creation of VideoEngine:
...
- clean a static variable just used to debug and not so necessary IMO.
- clean a really ugly reinterpret cast
- clean a extern "C" code and loading of dlls which is no longer in use.
Review URL: https://webrtc-codereview.appspot.com/1385006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3986 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-08 19:20:23 +00:00
andrew@webrtc.org
6155be2c61
Add /tools/protoc_wrappers to .gitignore.
...
TBR=pbos
Review URL: https://webrtc-codereview.appspot.com/1444004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3985 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-08 18:51:07 +00:00
phoglund@webrtc.org
aeb7d8757d
Tweaked webrtc_reformat.
...
Fixed variable names such as maskByte and stuff within brackets.
Fixed bug where we would think that for instance foo_internal.h was the self include when the right answer was foo.h.
Removed comment conversion: it was doing more damage than good.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1442005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3983 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-08 13:56:23 +00:00
phoglund@webrtc.org
315d39866e
Formatted dtmf_queue.
...
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1398004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3982 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-08 10:04:06 +00:00
kjellander@webrtc.org
73a4d5ab12
Add script to ensure virtual webcam is running.
...
This script will check that a webcam is running and start it if it's
not currently running.
It's tailored to the way our buildbots are currently configured.
TEST=local execution on Windows, Mac and Linux.
BUG=none
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1406005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3981 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-08 09:20:41 +00:00
pbos@webrtc.org
f6d67ae21f
Disable clang C++11 warnings to permit OVERRIDE keyword.
...
BUG=1623
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1431004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3980 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-08 08:34:34 +00:00
stefan@webrtc.org
d98e784f5f
Fix VCMProcessTimer::TimeUntilProcess() unsigned-integer underflow problem.
...
BUG=1665
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1341004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3979 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-08 06:38:53 +00:00
andrew@webrtc.org
b55a12ad32
Enable protobuf use in Chromium.
...
We might end up reverting this, but we need to get it committed and merged to
stable in order to test in a webrtc roll.
TBR=niklas.enbom
BUG=webrtc:830
Review URL: https://webrtc-codereview.appspot.com/1439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3978 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-08 00:03:30 +00:00
andrew@webrtc.org
e53084f837
Update protoc.gypi to match Chromium's latest.
...
This is in preparation for enabling protobufs in Chromium. Requires
syncing tools/protoc_wrapper.
BUG=webrtc:830
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1426004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3977 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 23:19:58 +00:00
niklas.enbom@webrtc.org
3be565b502
Refactoring for typing detection
...
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1370004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3976 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 21:04:24 +00:00
stefan@webrtc.org
ef14488d03
Trigger a PLI if the duration of non-decodable frames exceeds a threshold.
...
BUG=1663
R=mikhal@webrtc.org , ronghuawu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/1359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3975 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 19:16:33 +00:00
mikhal@webrtc.org
8f86cc8712
VCM/Receiver: Return null when can't extract frame.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1435004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3974 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 18:05:21 +00:00
mikhal@webrtc.org
474e915272
Relanding 3962: VCM/JB: Porting jitter_buffer_test to gtest
...
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1434004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3971 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 16:55:03 +00:00
mikhal@webrtc.org
759b041019
Relanding r3952: VCM: Updating receiver logic
...
BUG=r1734
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1433004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3970 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 16:36:00 +00:00
mikhal@webrtc.org
9c7685f9a6
VCM/JB: Break and skip to key if possible
...
BUG=1734
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1421004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3969 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 16:07:52 +00:00
pbos@webrtc.org
3004c79c6a
Fix clang errors in non-GYP_DEFINES=clang=1 build
...
BUG=1623
R=stefan@webrtc.org , tina.legrand@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1368004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3968 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 12:36:21 +00:00
stefan@webrtc.org
d3a1959678
Fix jitter buffer unittest.
...
TBR=mflodman@webrtc.org
BUG=1737
Review URL: https://webrtc-codereview.appspot.com/1430005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3967 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 12:35:58 +00:00
stefan@webrtc.org
a5dee33639
Correctly add packets to nack list when sequence number wraps.
...
BUG=1737
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1427004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3966 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 11:11:17 +00:00
pwestin@webrtc.org
0f29810288
Fix crash in pacer.
...
BUG=1731
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1410006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3964 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-06 16:37:22 +00:00
stefan@webrtc.org
4ce19b1664
Revert r3952 "VCM: Updating receiver logic"
...
TBR=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1410005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3963 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-06 13:16:51 +00:00
stefan@webrtc.org
273759048c
Revert r3956 "VCM/JB: Porting jitter_buffer_test to gtest."
...
TBR=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1408005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3962 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-06 13:12:58 +00:00
xians@webrtc.org
233c58de47
Landing 1399004, Minor clean up on the un-used _measureDelay code
...
Those code is/will never used, removing it makes the code better.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3961 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-06 11:52:47 +00:00
andrew@webrtc.org
59aaebc3cd
Add an option to override the TestToStderr trace printout time.
...
This is useful for offline file-based tests.
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1407004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3960 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-05 19:44:19 +00:00
andrew@webrtc.org
f9c289bafe
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
...
* Add the full license to all third party files.
* Correct some entries in LICENSE_THIRD_PARTY which were missing the full
license.
* Relicense all Chromium-licensed files under WebRTC.
* Remove third_party_mods/, which is now redundant.
R=jan.linden@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1396004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3959 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-05 18:54:10 +00:00
elham@webrtc.org
df3da84ec8
Updated WebRTC version number to 3.30
...
R=tnakamura@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1404005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3958 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-03 23:11:37 +00:00
mikhal@webrtc.org
45f2da0920
VCM/JB: Porting jitter_buffer_test to gtest.
...
Tests were not modified, but ported as is.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1391004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3956 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-03 22:22:46 +00:00
andrew@webrtc.org
a31c428307
Remove 44.1 kHz workaround from AudioDevice on PulseAudio.
...
We currently inform VoE that 44.1 kHz audio is 44 kHz. We now have arbitrary
resampling in VoE, allowing us to pass in the native 44.1 kHz.
Our ALSA interface always requires 48 kHz, allowing ALSA to handle resampling.
This also removes WEBRTC_PA_GTALK which was not defined anywhere.
BUG=webrtc:1395
TESTED=Using 44.1 for capture and render in loopback, ran through all codec channel/rate combinations. Quality is good. Testing AEC was difficult as I can't find a way to change the sample rate of an individual device in PulseAudio. Using a webcam at 32 kHz, other problems were the overriding contribution to quality degradation (delay issues, possible clock drift from the camera). At least I verified that the quality got no worse with this patch.
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1384004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3955 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-03 19:01:46 +00:00
andrew@webrtc.org
7cb766b016
Remove 44.1 kHz workaround from AudioDevice on WASAPI.
...
We currently inform VoE that 44.1 kHz audio is 44 kHz. We now have arbitrary
resampling in VoE, allowing us to pass in the native 44.1 kHz.
BUG=webrtc:1395
TESTED=Set capture device to 44.1 and render device to 48 and vice versa and observed good AEC. The quality is considerably worse before this change. Using 44.1 for capture and render in loopback, ran through all codec channel/rate combinations. Quality is good.
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1383004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3954 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-03 18:56:38 +00:00
sergeyu@chromium.org
bd4a2feddb
Fix off-by-one buffer overflow in WebRtcNetEQ_PacketBufferInsert().
...
BUG=1725
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1395004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3953 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-03 18:11:36 +00:00
mikhal@webrtc.org
d3cd565ecf
VCM: Updating receiver logic
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1363005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3952 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-03 17:54:18 +00:00
leozwang@webrtc.org
d293a58eaf
Correct and update dir name
...
TBR=jan.linden@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1403004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3950 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-03 17:16:40 +00:00
pbos@webrtc.org
77f6b2175e
Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..."
...
> Revert 3933 "Remove traces of deprecated WebRtc_Word types."
>
> > Remove traces of deprecated WebRtc_Word types.
> >
> > BUG=314
> > R=tommi@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/1385004
>
> TBR=pbos@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1386004
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1397004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3948 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-03 12:02:11 +00:00
solenberg@webrtc.org
2580bc4c30
Get rid of some unnecessary copying when sending REMBs.
...
BUG=
Review URL: https://webrtc-codereview.appspot.com/1325005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3947 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-03 09:22:14 +00:00
tina.legrand@webrtc.org
d5726a1286
Formatting ACM tests
...
Pure formatting of all files located in /webrtc/modules/audio_coding/main/test/
Smaller manual modifications done after using Eclipse formatting tool, like wrapping long lines (mostly comments).
BUG=issue1024
Review URL: https://webrtc-codereview.appspot.com/1342004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3946 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-03 07:34:12 +00:00
pwestin@webrtc.org
03efc89151
Fix when SetMinimumPlayoutDelay is configured to 0
...
BUG=1720
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1386005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3942 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 21:19:12 +00:00
pwestin@webrtc.org
42636e82d0
Removing bad code resulting in flaky test.
...
BUG=1723
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1390004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3941 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 21:02:04 +00:00
pwestin@webrtc.org
52b4e8871a
Adding trace and changing pacing constants
...
BUG=1721,1722
R=mikhal@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1380005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3940 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 19:02:17 +00:00
niklas.enbom@webrtc.org
a5961b855e
Update third party license file
...
R=jan.linden@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3939 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 18:49:28 +00:00
pwestin@webrtc.org
0d95e06a2f
Bugfix custom call stop.
...
BUG=1717
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1388004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3938 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 18:25:03 +00:00
andrew@webrtc.org
ea83c6ac9d
Allow voe_cmd_test to select Opus mono (now the default).
...
* Opus handles stereo and mono on the same payload type, so we need a different mechanism to choose between them.
* Assorted cleanups.
BUG=webrtc:1710
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3937 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 15:57:36 +00:00
andrew@webrtc.org
8c845cb623
Relax VoE's max packet length threshold.
...
The earlier threshold would cause packets from a currently available
codec (L16, 32 kHz, stereo) to be discarded.
TESTED=voe_cmd_test using L16, 32 kHz, stereo now works properly.
R=henrika@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1305008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3936 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 15:28:02 +00:00
phoglund@webrtc.org
258f55efc0
Disabled flaky test.
...
BUG=1719
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1387004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3935 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 12:35:00 +00:00
pbos@webrtc.org
68e5a68f07
Revert 3933 "Remove traces of deprecated WebRtc_Word types."
...
> Remove traces of deprecated WebRtc_Word types.
>
> BUG=314
> R=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1385004
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1386004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3934 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 09:30:12 +00:00
pbos@webrtc.org
265a5d298a
Remove traces of deprecated WebRtc_Word types.
...
BUG=314
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1385004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3933 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 09:11:20 +00:00
braveyao@webrtc.org
3c48f31e5b
WebRTCDemo Android app to route audio to headphone when it's plugged in.
...
BUG=1654
TEST=WebRTCDemo app
Review URL: https://webrtc-codereview.appspot.com/1348004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3932 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 03:18:00 +00:00
fbarchard@google.com
03d0c66376
Make libyuv fat on linux instead of thin.
...
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/1382004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3931 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 01:01:24 +00:00
andrew@webrtc.org
28e82bfec6
Replace Resampler with PushResampler in transmit_mixer.
...
* VoE can now exchange 44.1 kHz audio with AudioDevice.
* Changes still required in AudioDevice to remove the 44 kHz workarounds and
enable native 44.1 kHz.
BUG=webrtc:1395
TESTED=voe_cmd_test loopback running through codecs using all combinations of {8, 16, 32} kHz and {1, 2} channels, and Opus (48 kHz, stereo)
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1373004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3930 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 00:30:36 +00:00