generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
5,824 Commits 1 Branch 0 Tags 108 MiB
c70b2f9a54
Commit Graph

5824 Commits

This Branch
This Branch
All Branches
Author SHA1 Message Date
kwiberg@webrtc.org
f2e4a99a39 Add kwiberg@webrtc.org to watchlist for audio_coding and audio_processing 
BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6310 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 10:01:39 +00:00
buildbot@webrtc.org
b525a9d790 (Auto)update libjingle 68379861-> 68445177 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6309 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 09:42:15 +00:00
pbos@webrtc.org
044bdacfef Remove kMaxWaitForStatsMs from tsanv2 compilation. 
As some tests are #ifdef'd out on THREAD_SANITIZER this constant
triggers an unused-const-variable warning which breaks the build.

BUG=1205,3220
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6308 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 09:40:01 +00:00
kwiberg@webrtc.org
c0035a67a1 Remove an optimization that's no longer worth the extra complexity it causes 
The data_ optimization was a way to operate on the data directly
instead of copying it, applicable in the mono, non-float case. Since a
few audio_processing steps are already using floats (with more
hopefully to come), we don't end up benefiting from the optimization
anyway, so we might as well remove it.

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

Review URL: https://webrtc-codereview.appspot.com/15539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6307 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 09:10:06 +00:00
buildbot@webrtc.org
34a08b4fb8 (Auto)update libjingle 68275107-> 68379861 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6305 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 15:48:10 +00:00
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
kjellander@webrtc.org
2f7c7ce020 Remove old perf_expectations no longer used. 
This has been replaced with the Chromium Perf
Dashboard web application a long time ago.

BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6303 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 15:03:21 +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
pbos@webrtc.org
174a67439b Enable -Wall, -Wextra and -Wunused-variable for talk/ on clang. 
Also removes one case of unused-variable.

BUG=3220
R=henrike@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6297 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 07:58:30 +00:00
jiayl@webrtc.org
8a09af3f67 Fix the build error from OpenSSLStreamAdapter::SSLVerifyCallback 
TBR=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/17639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 23:24:08 +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
jiayl@webrtc.org
0163674f99 Make OpenSSLStreamAdapter verify the leaf certificate digest for chained certificates. 
It used to compre a parent certificate's digest against the SDP fingerprint and caused connection failure.

BUG=3383
R=bemasc@webrtc.org, juberti@webrtc.org, rsleevi@chromium.org

Review URL: https://webrtc-codereview.appspot.com/17589005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6294 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 23:14:08 +00:00
andrew@webrtc.org
222d8d3b1f Add a TSAN suppression for a benign TRACE_EVENT race. 
TBR=hclam
BUG=3409

Review URL: https://webrtc-codereview.appspot.com/15639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 23:14:00 +00:00
tkchin@webrtc.org
56d114627b Fix AppRTC target configuration in libjingle_examples.gyp. 
libjingle_peerconnection_objc doesn't exist as a target in 32bit, so AppRTCDemo
needs that guard as well.

R=andrew@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/18489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6292 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 23:04:39 +00:00
tkchin@webrtc.org
acca675bcf Implement mac version of AppRTCDemo. 
- Refactored and moved AppRTCDemo to support sharing AppRTC connection code between iOS and mac counterparts.
- Refactored OpenGL rendering code to be shared between iOS and mac counterparts.
- iOS AppRTCDemo now respects video aspect ratio.

BUG=2168
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6291 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 22:26:06 +00:00
jiayl@webrtc.org
9f8164c060 Fix two bugs in DataChannel state transition. 
1. OnStateChange should not be fired if state is not changed.
2. RemotePeerRequestClose should be a no-op if it's already closed.

TBR=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/21559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 21:53:17 +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
buildbot@webrtc.org
1678db9df6 (Auto)update libjingle 68230113-> 68244456 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6287 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 14:02:09 +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
pbos@webrtc.org
b4e3c254ee Add native_test dependency to webrtc_perf_tests. 
Required to run the binary on Android bots.

BUG=3423
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6285 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 12:42:10 +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
buildbot@webrtc.org
540a2251aa (Auto)update libjingle 68230011-> 68230113 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6281 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 07:40:35 +00:00
pbos@webrtc.org
35efb839ed Implement new-API test RecvStreamWithoutRtx. 
R=pthatcher@google.com, pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/20449005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 07:40:04 +00:00
pbos@webrtc.org
c34bb3a886 Log default receive stream creation. 
Log when receiving a packet that doesn't have a receiver, this way you
can tell from logs where the AddRecvStream call came from.

R=pthatcher@google.com, pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/17459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6279 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 07:38:43 +00:00
pbos@webrtc.org
198647473b Implement and fix new-API NackIsEnabled test. 
Required enabling NACK on receiver side which was apparently missed.

BUG=1788
R=pthatcher@google.com, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16499007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6278 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 07:35:47 +00:00
buildbot@webrtc.org
1d66be22c8 (Auto)update libjingle 68203780-> 68206793 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 22:54:24 +00:00
jiayl@webrtc.org
8dcd43c4f7 Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF. 
This is the first step toward switching completely to UDP/TLS/RTP/SAVPF.

BUG=2796
R=juberti@webrtc.org, pthatcher@google.com

Review URL: https://webrtc-codereview.appspot.com/13439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6276 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 22:07:59 +00:00
fischman@webrtc.org
abe01dd634 AppRTCDemo(android): run in full-screen & immersive mode. 
Also:
- Only show stats HUD on demand
- Only collect stats when HUD is showing
- Don't render solid green frame when video is not present in either direction

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6275 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 21:46:52 +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
wu@webrtc.org
7a9a3b70b3 * Revert clock.cc changes made in 6178, but keep the changes to the test. 
* Use the new appoach proposed by jib in https://review.webrtc.org/10439004/ to fix the windows clock issue.

BUg=3325

R=niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15569005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6273 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 19:40:28 +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
henrike@webrtc.org
caa01b172e Rebase webrtc/base with r6250: 
cd webrtc/base
svn diff -r 6249:6250 http://webrtc.googlecode.com/svn/trunk/talk/base >
6250.diff
patch -p0 -i 6250.diff

BUG=3379
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6271 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 15:53:39 +00:00
jiayl@webrtc.org
5dc51fbe50 Closes the DataChannel when the send buffer is full or on transport errors. 
As stated in the spec.

BUG=2645
R=pthatcher@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6270 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 15:33:54 +00:00
jiayl@webrtc.org
001fd2d503 Fire OnRenegotiationNeeded only for the first SCTP DataChannel. 
Subsequent DataChannels do not need renegotiation since SCTP data streams are not negotiated through SDP.

BUG=2431
R=pthatcher@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6268 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 15:31:11 +00:00
wu@webrtc.org
9aa7d8df95 Increase the threshold for CallPerfTest.CaptureNtpTimeWithNetworkDelay to avoid flaky. 
BUG=3374
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6267 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 05:03:52 +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
fischman@webrtc.org
43a1395370 AppRTCDemo(android): README updates for a shrinking envsetup.sh world. 
There was duplicated (and out of date!) information in README relative to
getting-started so de-duped to point to getting-started as the canonical
reference.

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15589006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6265 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 17:29:09 +00:00
jiayl@webrtc.org
b364016cbb Revert r6161 "Drop the DataChannel message if it's received when the channel is not open." 
The spec does not say the DataChannel has to be open to receive a message.

TBR=pthatcher@google.com
BUG=crbug/363005

Review URL: https://webrtc-codereview.appspot.com/16569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6264 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 16:37:25 +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
c1a40a7b68 This CL is to adding feedback of packet loss rate to encoder in voice engine. A direct reason for doing it is to make use of Opus FEC, which can adapt itself to changes in the packet loss rate. 
This CL is going to be combined with another CL in ACM, which is to be landed.

TEST=passed_try_bots
BUG=
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6262 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 09:52:06 +00:00
bjornv@webrtc.org
aca5939dfc common_audio/signal_processing: Fixes arm compilation issues with gcc 4.8 
In r6240 gcc was rolled from 4.6 to 4.8 changing the behavior on arm. The output of ComplexFFT differs causing both AECM and NS to perform worse. Looking at issues on gcc it says that there could be a memory shuffling/optimization despite using volatile affecting the output.
Splitting the three instructions in one call into two separate calls makes the compiler take proper actions resulting in correct outputs.

BUG=3370,3395
TESTED=trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6261 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 08:45:04 +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