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Author SHA1 Message Date
solenberg@webrtc.org
a28c697d93 - Get rid of 'using' from .h 
- Add parenthesis to make order of evaluation clearer.

BUG=
R=minyue@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6304 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 15:22:33 +00:00
henrik.lundin@webrtc.org
2bd032e11c Disable MouseCursorMonitorTest 
Last attempt reverted. Trying again in a different way.

This CL effectively reverts r6300.

BUG=3245
TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/20549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6301 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 14:52:34 +00:00
henrik.lundin@webrtc.org
4ecae6e753 Disable MouseCursorMonitorTest.FromScreen 
The test is flaky.

BUG=3245
TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/21579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6300 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 14:17:06 +00:00
henrik.lundin@webrtc.org
fe41a8f68d Adding thread annotations to parts of Audio Coding Module 
Picking some low-hanging fruit. Add annotations for acm_crit_sect_ that
do not require lock changes. Also adding annotations for callbacks.

BUG=3401
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6299 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 11:45:26 +00:00
bjornv@webrtc.org
2812b59acd Re-enables CommonFormats test for Android. 
It seems like this was a one time only and not a flaky test.

BUG=3376
TESTED=trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15649005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6298 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 11:27:29 +00:00
fischman@webrtc.org
360507b12b VideoCaptureAndroid: don't synchronized on camera thread. 
BUG=3421
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6295 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 23:17:38 +00:00
andrew@webrtc.org
1fddd6185d Add a Reset() method to AudioFrame. 
This method is introduced to try to avoid inconsistent resetting of
AudioFrame members to default/uninitialized values.

Use it at the call points of DownConvertToCodecFormat(). Results in the
following minor functional changes:
- speech_activity_ is set to its uninitialized value. AFAICT, this
member isn't used at all in the capture path.
- timestamp_ is switched from -1 to 0. This member doesn't appear to be
used either in the capture path, but left a TODO for wu to change the
default value to better represent the uninitialized state.

Bonus: Don't copy the frame on error in RemixAndResample(). An error
indicates a logical fault (as pointed out by the asserts) that we should
not attempt to recover from.
BUG=3111
R=turaj@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21519007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6289 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 17:28:50 +00:00
andrew@webrtc.org
af48aaadf4 Disable AudioCodingModuleMtTest due to memcheck and tsan failures. 
This is a new test; the failures are not due to a change in underlying code.

TBR=henrik.lundin
BUG=3426

Review URL: https://webrtc-codereview.appspot.com/19589005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6288 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 17:11:15 +00:00
henrik.lundin@webrtc.org
288bd15db8 Multi-threaded test for Audio Coding Module 
This CL adds a basic multi-threaded extention of the ACM unit test.
The test has three threads. One thread adds raw audio to the sender
side and encodes it. The next thread adds encoded RTP packets to the
receiver. The last thread pulls decoded audio out of the receiver.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6286 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 13:00:35 +00:00
stefan@webrtc.org
420b2567f3 Fix bug where RTP headers in the packet history were replaced with the RTX wrapped headers. 
This caused only the first retransmission to be successful.
Introduced with https://code.google.com/p/webrtc/source/detail?r=5728.

BUG=1811
R=asapersson@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 12:17:15 +00:00
minyue@webrtc.org
a816180f93 Fixing a bug regarding VOE packet loss rate feedback to ACM 
Phenomenon:

When packet loss rate was fed to a codec that does not implement packet loss adaptive encoding, VoE logs an error.

Reason:

The ACM function SetPacketLossRate(int rate) return -1 unnecessarily too often. It was intended for more severe errors like
1. codec is not ready
2. input rate is out of range

BUG=webrtc:3413
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 09:28:07 +00:00
sprang@webrtc.org
6e732c6765 Revert 6272 "Update generated asm offsets scripts." 
Revert since it fails webrtc-in-chromium Android bots.

> Update generated asm offsets scripts.
>
> Libvpx updated the unpack scripts to fix building dependencies.
>
> Roll libvpx 269083:273304
> See https://codereview.chromium.org/295313002/
> https://codereview.chromium.org/298063002/
> https://codereview.chromium.org/305533008/
> https://codereview.chromium.org/305703002/
> https://codereview.chromium.org/298383003/
> https://codereview.chromium.org/302863004/
> for the libvpx changes.
>
> BUG=377062
> R=andrew@webrtc.org, michaelbai@chromium.org
>
> Review URL: https://webrtc-codereview.appspot.com/12579008

TBR=fgalligan@google.com

Review URL: https://webrtc-codereview.appspot.com/12649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6282 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 09:19:03 +00:00
wu@webrtc.org
21a5d449b7 Increase VPMVideoDecimator's initial max_frame_rate_ to 60, which allow us potentially do 60fps. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21499006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6274 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 19:43:26 +00:00
fgalligan@google.com
2a8efa8971 Update generated asm offsets scripts. 
Libvpx updated the unpack scripts to fix building dependencies.

Roll libvpx 269083:273304
See https://codereview.chromium.org/295313002/
https://codereview.chromium.org/298063002/
https://codereview.chromium.org/305533008/
https://codereview.chromium.org/305703002/
https://codereview.chromium.org/298383003/
https://codereview.chromium.org/302863004/
for the libvpx changes.

BUG=377062
R=andrew@webrtc.org, michaelbai@chromium.org

Review URL: https://webrtc-codereview.appspot.com/12579008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6272 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 17:08:34 +00:00
fischman@webrtc.org
d6a0efdc86 VideoCaptureAndroid: quit & join the camera thread on stopCapture. 
Also fix latent bug where setPreviewRotation() wouldn't hold
the lock while its delegate setPreviewRotationOnCameraThread()
was running, allowing the camera to be freed between the
null-check and the use.

BUG=3389
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6266 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 18:37:07 +00:00
kwiberg@webrtc.org
f15c14be22 Echo canceler: Saturate output to guarantee it'll be in the allowed range 
r6138 (https://webrtc-codereview.appspot.com/18399005/) somewhat
ill-advisedly removed the saturation step at the end of
aec_core.c:NonLinearProcessing(); this patch restores it.

BUG=
R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6263 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 11:47:08 +00:00
minyue@webrtc.org
0aa3ee661c Better buffer size estimation in NetEq for redundant packets 
TEST=passed_all_trybots
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 07:48:01 +00:00
henrik.lundin@webrtc.org
1b9df05c85 Revert 6257 "Rename neteq4 folder to neteq" 
> Rename neteq4 folder to neteq
> 
> BUG=2996
> R=turaj@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/12569005

TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6259 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 07:33:39 +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
henrik.lundin@webrtc.org
a90f6d67f7 Rename neteq4 folder to neteq 
BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12569005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6257 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 06:23:34 +00:00
andrew@webrtc.org
27e884cf47 Disable MouseCursorMonitorTest due to flake on Windows. 
TBR=sergeyu
BUG=3408

Review URL: https://webrtc-codereview.appspot.com/15589005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6256 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 03:34:04 +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
henrik.lundin@webrtc.org
74767401f2 Fix a bug preventing FilePlayer from playing encoded wav files 
A bug in ACM2 prevented decoding and playout of wav files where the
audio data was encoded (i.e., not just linear PCM 16 bit data).

This CL fixes the issue, and adds a unit test for the FilePlayer.

BUG=3386
R=henrike@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21499005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6248 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-26 13:37:45 +00:00
turaj@webrtc.org
546961a9d3 Avoid reading uninitialized values (outside baundary) in DFT arithmatic decoder of iSAC-fix. 
Arithmetic encoder does not right the last 2 or 3 bytes of |streamval| when terminating the bit-stream. Perhaps the last bytes makes no difference in decoding the stream. However, the decoder reads full |streamval| (int16_t) going out of boundary and reading uninitialized values. This avoids this problem. by inserting zero-bytes whenever decoder intends to read outside boundary.

BUG=1353,chrome373312,b/13468260
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16499005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6234 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 17:14:29 +00:00
minyue@webrtc.org
aa5ea1c0f9 1. Make a clear distinction between codec internal FEC and RED, confusing mentioning of FEC in the old codes is replaced by RED 
2. Add two new APIs to configure codec internal FEC

3. Add a test and listened to results. This is based modifying EncodeDecodeTest and deriving a new class from it.

New ACM gives good result.
Old ACM does not use NetEq 4, so FEC won't be decoded.

BUG=
R=tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6233 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 15:16:51 +00:00
henrike@webrtc.org
1bb5da04fe Adds missing include of assert header. 
BUG=3380
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/14569008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6221 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-22 14:31:14 +00:00
braveyao@webrtc.org
21f7d6d2fe WebRTCDemo: move the deletion of CritSect to end of the dtor to fix a crash in Android video renderer. 
BUG=3368
TEST=Manual Test

Review URL: https://webrtc-codereview.appspot.com/21519005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6220 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-22 02:57:55 +00:00
henrike@webrtc.org
88fbb2d86b Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. 
Same as https://webrtc-codereview.appspot.com/19519004. The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing...
(tested locally).

BUG=3380
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 21:18:46 +00:00
jiayl@webrtc.org
7ca277b574 Initializes WINDOWPLACEMENT::length in GetCroppedWindowRect. 
BUG=https://code.google.com/p/webrtc/issues/detail?id=3196
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/21529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6213 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 16:02:31 +00:00
mcasas@webrtc.org
2fa7f79094 Revert 6202 "Switch to using base/constructormagic.h and remove ..." 
> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
> 
> BUG=N/A
> R=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/19519004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 11:07:29 +00:00
mcasas@webrtc.org
c2213b6a0f Revert 6208 "Patch from henrike@webrtc.org" 
Wasn't enough. I'll have to revert the whole rev 6202.

> Patch from henrike@webrtc.org
> https://code.google.com/p/webrtc/source/detail?r=6202
> didn't work for at least one file and broke most of 
> the compile steps in the FYI bots. The file is reverted
> here.
> 
> TBR= henrike@webrtc.org, sergeyu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/17609004

TBR=mcasas@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6209 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 10:03:09 +00:00
mcasas@webrtc.org
86df8acc92 Patch from henrike@webrtc.org 
https://code.google.com/p/webrtc/source/detail?r=6202
didn't work for at least one file and broke most of 
the compile steps in the FYI bots. The file is reverted
here.

TBR= henrike@webrtc.org, sergeyu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6208 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 08:40:56 +00:00
henrik.lundin@webrtc.org
48438c2c90 Enabling NetEq bit-exactness test for Win x64 
A new reference file (neteq4_universal_ref_win_64.pcm) was generated and
uploaded.

Also removing the old hack to have different reference files
for different version of Visual Studio. The test is now only supporting
VS 2012 and later (_MSC_VER >= 1700). This makes the windows 32-bit
output identical to the generic reference file
(neteq4_universal_ref.pcm), so the specialized one
(neteq4_universal_ref_win_32.pcm) could have been removed. However,
since the resources sync mechanism does not include removing of old
files, a client could pick up the old reference and fail. Therefore,
this cl also updates neteq4_universal_ref_win_32.pcm to be identical to
neteq4_universal_ref.pcm.

BUG=1458
R=kjellander@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14569005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6204 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 16:07:43 +00:00
henrike@webrtc.org
125ffd709d Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 15:20:44 +00:00
stefan@webrtc.org
70bb2d5755 Revert r6198 "Expose the original packet length in in the RTP play tools." 
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6200 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 13:25:48 +00:00
stefan@webrtc.org
e208458643 Expose the original packet length in in the RTP play tools. 
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6198 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 13:09:16 +00:00
stefan@webrtc.org
be4ab99a53 Disabling RealFFTTest.RealAndComplexMatch and AudioProcessingTest.Formats as they currently are broken with gcc 4.8. 
BUG=3370
R=bjornv@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6197 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 12:42:01 +00:00
henrik.lundin@webrtc.org
a36db970bd Suppress GMOCK printouts from TestVideoSenderWithVp8 
Adding a missing EXPECT_CALL.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6196 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 11:16:10 +00:00
asapersson@webrtc.org
a826006132 Add NACK and RPSI packet types to RTCP packet builder. 
Fixes bug found when parsing received RPSI packet.

BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6194 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 09:53:51 +00:00
wu@webrtc.org
cb711f77d2 Add interface to propagate audio capture timestamp to the renderer. 
BUG=3111
R=andrew@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6189 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 17:39:11 +00:00
pbos@webrtc.org
ebb467fdc8 Avoid NACK-list flush error on keyframe packets. 
Receiver code used to indicate a flush error even if the incoming packet
is a keyframe, forcing a request of a keyframe. Now it takes this
keyframe into account and doesn't error as the stream is decodable from
this point.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15549005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 15:28:02 +00:00
stefan@webrtc.org
64339a7069 Don't crash if a frame returned from the decoder is too old. 
BUG=crbug/371805
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6187 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 13:31:35 +00:00
michaelbai@google.com
725e582461 Use the new gyp_var_prefix local variable set by gyp instead of the 
global GYP_VAR_PREFIX set by the makefiles, since the latter is not
guaranteed to still be the same value at the time the command is
executed. Also, use abspath instead of realpath to convert paths to
absolute, since realpath expands to the empty string if the target file
doesn't exist, complicating build debugging.

BUG=
R=andrew@webrtc.org, torne@chromium.org

Review URL: https://webrtc-codereview.appspot.com/12559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6186 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 17:56:10 +00:00
henrike@webrtc.org
14abcc7322 libvpx's UNUSED macro conflicts with webrtc/base's. Added missing include of assert.h. Globally defined function "Unused" in talk/base and its copy (webrtc/base) is causing a conflict. 
libvpx macro (UNUSED) can be found here:
http://src.chromium.org/viewvc/chrome/trunk/deps/third_party/libvpx/source/libvpx/vpx/vpx_codec.h

BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6185 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 16:54:44 +00:00
bjornv@webrtc.org
a3b5673879 common_audio/signal_processing: Removes macro WEBRTC_SPL_UMUL_RSFT16 
This macro was only used on two lines in iSACfix and I replaced those with the operations the macro performed.

BUG=3348
TESTED=trybots, manual unittests
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 12:11:20 +00:00
bjornv@webrtc.org
1b21a57902 common_audio/signal_processing: Removed macro WEBRTC_SPL_SUB_SAT_W16 
Macro was only mapping a function used in one place.

BUG=3348
TESTED=trybots, unittests
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6180 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 06:40:31 +00:00
mflodman@webrtc.org
d5da25063c Revert "Revert "Audio processing: Feed each processing step its choice 
of int or float data"

This reverts commit 6142.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6172 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 11:17:21 +00:00
henrik.lundin@webrtc.org
b4e80e095f Re-enable almost all NetEqDecodingTests for Android 
All but three tests in NetEqDecodingTest could be re-enabled without
any changes. Also making sure that the TestNetworkStatistics test exits
on first diff. (Otherwise, the log output gets flooded with error
messages.)

The tests that are still disabled are:
NetEqDecodingTest.TestBitExactness
NetEqDecodingTest.TestNetworkStatistics
NetEqDecodingTest.DecoderError

BUG=3343
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6168 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 07:14:00 +00:00
braveyao@webrtc.org
7cb4752184 WebRTCDemo: couldn't run a second time. The reason is voe could register/unregister for each run, but vie would expect initialization only once per process. 
This cl is to teach videocapture android how to deinitialize and allow it to be re-initializable.

BUG=3284
TEST=ManualTest
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6167 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 03:18:15 +00:00