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Author SHA1 Message Date
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
stefan@webrtc.org
4980679d35 Fixes two bugs in receive statistics. 
- Reported bitrate wasn't reset correctly when no frames had been received.
- Internal framerate estimate wasn't reset when no frames had been received.

BUG=1713

Review URL: https://webrtc-codereview.appspot.com/1377004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3924 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 22:05:07 +00:00
pwestin@webrtc.org
d35964a1ce Fixing AV sync. 
Increased 2 const to allow for a bigger difference in AV sync.

BUG=1711

Re-wrote the ComputeDelays to be readable and remove the possibilities of returning values lower than base_target_delay_ms

R=mflodman@webrtc.org, mikhal@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1367004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3922 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 16:06:10 +00:00
mikhal@webrtc.org
6faba6edc9 VCM: Setting buffering delay in timing 
Review URL: https://webrtc-codereview.appspot.com/1338004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3921 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 15:39:34 +00:00
mikhal@webrtc.org
dd807ac474 Adding buffered mode to loopback test 
Review URL: https://webrtc-codereview.appspot.com/1371004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3920 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 15:19:47 +00:00
solenberg@webrtc.org
8efc623fc2 Apply Chromium C++ style to RemoteRateControl. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3919 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 08:33:46 +00:00
sergeyu@chromium.org
15e32ccd30 Add DesktopCapturer interface for desktop capturers. 
The new DesktopCapturer interface will be used for screen and window
captures. Beside DesktopCapturer itself also added classes/interfaces
that it depends on.

R=alexeypa@chromium.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1322007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3917 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-29 20:10:57 +00:00
mikhal@webrtc.org
865ada3a52 Don't reset the last je value and mode 
Review URL: https://webrtc-codereview.appspot.com/1369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3916 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-29 19:09:41 +00:00
andrew@webrtc.org
50b2efef6e Add a wrapper around PushSincResampler and the old Resampler. 
The old resampler is used whenever it supports the requested rates. Otherwise
the sinc resampler is enabled.

Integrated with output_mixer in order to test the change through
output_mixer_unittest. The sinc resampler will not yet be used, since we don't
feed VoE with any rates that trigger it.

BUG=webrtc:1395
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1355004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3915 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-29 17:27:29 +00:00
stefan@webrtc.org
5b7120c81b Fix two issues where we might end up busy looping in decoder_render mode. 
This happens if
- Next frame is far into the future (> 200 ms).
- Next frame is ready for decode/render but incomplete.

BUG=1696
TESTS=trybots

Review URL: https://webrtc-codereview.appspot.com/1354005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3914 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-29 16:41:30 +00:00
pwestin@webrtc.org
b0061f94b2 Enable Nack pacing. 
Review URL: https://webrtc-codereview.appspot.com/1357004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3912 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-27 00:41:08 +00:00
mikhal@webrtc.org
47128ab5ab Removing vie file related code from vie_custom_call 
Follow up on https://code.google.com/p/webrtc/source/detail?r=3900

Review URL: https://webrtc-codereview.appspot.com/1361004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3911 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-26 20:09:54 +00:00
pwestin@webrtc.org
4e545b33b3 Fixed remaining nits from Stefan 
Review URL: https://webrtc-codereview.appspot.com/1323007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3910 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-26 15:23:34 +00:00
andrew@webrtc.org
8fc05feed4 Add a push-based wrapper around SincResampler. 
Includes a unittest to ensure we meet the same quality thresholds as SincResampler (modulo quantization error).

BUG=webrtc:1395

Review URL: https://webrtc-codereview.appspot.com/1323011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3909 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-26 14:56:51 +00:00
fbarchard@google.com
42b0b84367 libyuv r680 fixes arm version of I444ToARGB and some lint changes 
BUG=none
TEST=libyuv unittests pass on arm with Neon disabled.
Review URL: https://webrtc-codereview.appspot.com/1356005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3908 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-26 02:23:32 +00:00
andrew@webrtc.org
1acb3b33bc Add comfort noise disabling and routing mode selection to audioproc. 
TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/1358004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3907 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-26 00:39:27 +00:00
mikhal@webrtc.org
4cea79b830 Removing another instance of file api 
Review URL: https://webrtc-codereview.appspot.com/1356004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3906 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 23:31:27 +00:00
vikasmarwaha@webrtc.org
77ac84814d Added new demo states.html & updated existing demos to work on firefox. 
Review URL: https://webrtc-codereview.appspot.com/1327007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3905 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 23:22:03 +00:00
pwestin@webrtc.org
91563e42da Fix the encoder pause logic. 
BUG=1691

Review URL: https://webrtc-codereview.appspot.com/1352004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3904 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 22:20:08 +00:00
mikhal@webrtc.org
381da4be9c VCM: Adding API for the size(duration) of the jitter buffer. 
Refers to the duration in time of the frames which are ready to be sent to the decoder.

Review URL: https://webrtc-codereview.appspot.com/1319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3903 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 21:45:29 +00:00
mikhal@webrtc.org
8392cd9edd VCM/JB: Using last decoded state for waiting for key 
relanding 1323006

BUG=

Review URL: https://webrtc-codereview.appspot.com/1354004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3902 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 21:30:50 +00:00
mikhal@webrtc.org
dc3cd217b2 VCM/JB: FrameForDecoding->IncompleteFrameForDecoding 
- Update complete frame for decoding
- Remove FrameForDecodingNack

This CL should only be committed after issue http://webrtc-codereview.appspot.com/1313007/

Review URL: https://webrtc-codereview.appspot.com/1316007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3901 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 20:27:04 +00:00
mikhal@webrtc.org
b84f13f185 Disabling avi file interface 
Review URL: https://webrtc-codereview.appspot.com/1351004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3900 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 18:07:32 +00:00
pwestin@webrtc.org
52aa019e98 Avoid adding duplicates in pacer lists. 
Review URL: https://webrtc-codereview.appspot.com/1329007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3899 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 17:35:56 +00:00
stefan@webrtc.org
cb60fb2e6c Make sure timestamps are monotonically increasing. 
BUG=1563

Review URL: https://webrtc-codereview.appspot.com/1325008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3898 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-24 23:54:31 +00:00
fbarchard@google.com
c63772eb39 libyuv license file updates for Android WebView license check. 
BUG=none
TEST=try bots
Review URL: https://webrtc-codereview.appspot.com/1313013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3897 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-24 02:12:49 +00:00
andrew@webrtc.org
df9c0e5ec9 Revert 3892 "VCM/JB: Using last decoded state for waiting for key" 
> VCM/JB: Using last decoded state for waiting for key
> 
> Review URL: https://webrtc-codereview.appspot.com/1323006

Although I have no idea why, it appears this might be causing failures in ViEStandardIntegrationTest.RunsFileTestWithoutErrors. I was unable to reproduce locally. This is a trial revert to verify. If the errors continue to happen, I will restore this change.

TBR=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1321010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3896 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-24 02:13:18 +00:00
andrew@webrtc.org
2e65346e98 Add a root codereview.settings file. 
Chromium has one of these as well:
http://src.chromium.org/viewvc/chrome/codereview.settings

It's needed for drover reverts to work properly.

TBR=kjellander

Review URL: https://webrtc-codereview.appspot.com/1319014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3895 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-24 02:08:50 +00:00
fbarchard@google.com
6fec15941a Update template to follow chromium copyright style 
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/1322008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3894 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-24 01:01:28 +00:00
andresp@webrtc.org
b5eeaa92ba Adding extra options to interact with external encoder/decoder. 
Review URL: https://webrtc-codereview.appspot.com/1327006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3893 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 22:50:53 +00:00
mikhal@webrtc.org
1248d4effc VCM/JB: Using last decoded state for waiting for key 
Review URL: https://webrtc-codereview.appspot.com/1323006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3892 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 20:57:06 +00:00
fbarchard@google.com
3a9a3cdaa2 Roll libyuv to r676 for improved llvm compatibility 
BUG=none
TESTED=try bots
Review URL: https://webrtc-codereview.appspot.com/1313010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3891 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 18:05:46 +00:00
stefan@webrtc.org
8ca8a71de2 Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." 
This reverts commit aae26db1da5803482b094357c546b8454ab1c26d.

BUG=1613
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1327008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3890 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 16:48:32 +00:00