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Author SHA1 Message Date
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
andrew@webrtc.org
342353780d Consolidate common_audio into a single target. 
In principle should reduce gyp processing time, but the difference was not measurable. In any case, it's a good simplification that aligns with having a single common_video target.

R=bjornv@webrtc.org, kma@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1375004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3928 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 23:43:26 +00:00
andrew@webrtc.org
dff69c56b0 Add AEC suppression level option to audioproc. 
TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/1368007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3927 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 23:01:09 +00:00
sergeyu@chromium.org
23516638fa Move WEBRTC_THREAD_RR and WEBRTC_CLOCK_TYPE_REALTIME to system_wrappers.gypi . 
WEBRTC_THREAD_RR and WEBRTC_CLOCK_TYPE_REALTIME are used only in code compiled
in system_wrappers, so they don't need to be in common.gypi.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1368005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3926 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 22:53:51 +00:00
andresp@webrtc.org
72d0b0cf1f Add self to video_engine watchlist. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1305009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3925 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 22:20:53 +00:00