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
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
marpan@webrtc.org
0ef565ee7d
Roll libvpx 267596:269083
...
R=andrew@webrtc.org
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6255 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-27 21:13:52 +00:00
fischman@webrtc.org
033aa2217d
video_engine_tests_apk: enable running by adding nativeRunTests dependency.
...
BUG=2462
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12579007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6254 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-27 18:44:59 +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
59336e85fb
Adding R/W lock to SimulatedClock
...
This change makes the SimulatedClock class thread safe.
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6251 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-27 09:34:58 +00:00
phoglund@webrtc.org
f666ecc60d
Disabling flaky libjingle tests after fixit week.
...
BUG=webrtc:3316,webrtc:3317,webrtc:3318
TBR=fischman@google.com
Review URL: https://webrtc-codereview.appspot.com/12569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6250 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-27 08:08:00 +00:00
asapersson@webrtc.org
ab6bf4f54c
Added api for getting cpu measures using a struct.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6249 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-27 07:43:15 +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
asapersson@webrtc.org
1457b4737a
First incoming packet was not accounted for in receive stats. Changed call order for incoming packet to receive statistics class.
...
Receive stats is reset if the payload type changes. Update stats after a possible reset.
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6247 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-26 13:06:04 +00:00
buildbot@webrtc.org
727ff69829
(Auto)update libjingle 67872893-> 67873348
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6244 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 23:20:53 +00:00
buildbot@webrtc.org
75cb3dc5f2
(Auto)update libjingle 67869540-> 67872893
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6243 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 23:13:35 +00:00
mallinath@webrtc.org
b445f26f24
Fixing correct UMA metric for PeerConnection enabled with IPv4 Vs IPv6.
...
BUG=N/A
TBR=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21499007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6242 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 22:19:37 +00:00
fischman@webrtc.org
440e1d1053
vie_autotest_android.cc: stop referring to undefined functions.
...
The roll in r6240 exposed the fact that vie_autotest_android.cc has been
depending on vie_autotest_network.cc since forever, even though that file isn't
part of the build! #if'ing the references out to green the build.
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17599005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6241 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 21:40:45 +00:00
fischman@webrtc.org
4610f1d427
Roll chromium_revision 266514:272489
...
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6240 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 20:56:12 +00:00
henrike@webrtc.org
ddc79d0418
Rebase webrtc/base with r6232:
...
cd webrtc/base
svn diff -r 6231:6232 http://webrtc.googlecode.com/svn/trunk/talk/base > 6232.diff
patch -p0 -i 6232.diff
BUG=3379
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6239 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 18:40:46 +00:00
fischman@webrtc.org
39eccefbde
Disable ChannelManagerTest.StartupShutdownOnUnstartedThread
...
The test is testing a scenario that shouldn't happen.
BUG=3388
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21509005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6238 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 17:50:38 +00:00
buildbot@webrtc.org
7aa1a4767f
(Auto)update libjingle 67848628-> 67848776
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6237 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 17:33:05 +00:00
fischman@webrtc.org
e5063b1733
Thread: delete racy API (Release()) and fix racy code (started()).
...
- Thread::Release() wrote a local variable on the calling thread but read it on
another thread, with no synchronization. Happily it has no non-test callers
so deleting it instead of trying to fix it (see bug for details).
- Thread::started_ similarly was racily being written to; replaced with a
running_ Event, and hid the accessor except for tests & legacy callers,
with a note about why it's a bad idea.
webrtc/base patched with:
git diff origin --relative=talk/base | patch -p1 -dwebrtc/base
followed by manual merge of 3 thunks that ran afoul of naming differences
between talk/base and webrtc/base.
BUG=3388
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14589005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6236 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 17:28:50 +00:00
fischman@webrtc.org
18f41b8eb4
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
...
Example files that this makes ok instead of flagging include:
talk/base/signalthread_unittest.cc
talk/base/thread_unittest.cc
webrtc/base/signalthread_unittest.cc
webrtc/base/thread.cc
webrtc/base/thread.h
webrtc/base/thread_unittest.cc
BUG=1027
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19539006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6235 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 17:27:18 +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