mflodman@webrtc.org
c320027d6a
Don't print a warning if RTPPacketHistory::SetStorePacketStatus is called
...
twice with the same settings.
Without this change, setting up a call with the new video API will
print a trace warning.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5566 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 14:51:00 +00:00
turaj@webrtc.org
2086e0fbf3
Remove unnecessary warnings.
...
BUG=
TEST=try job
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8719005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5565 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 14:22:20 +00:00
solenberg@webrtc.org
a07923339b
Remove external encryption API for VoE.
...
BUG=
R=henrika@webrtc.org , henrikg@webrtc.org , phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5564 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 11:27:22 +00:00
kjellander@webrtc.org
0a9d822812
Change mime type to text/html for multiple-relay.html
...
R=hta@chromium.org
Review URL: https://webrtc-codereview.appspot.com/8809005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5563 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 08:45:13 +00:00
sprang@webrtc.org
346094cb01
Incorrect overhead calculation when using FEC + RTP extension headers.
...
When frames are fragmented inte multiple RTP packets in order to not
exceed a maximum packet size, the header overhead calculation must
take into account that FEC redundancy packets may use more than the
12 bytes of the basic RTP header. For example, a csrc list or extension
headers may be present.
BUG=2899
R=phoglund@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8769005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5562 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 08:40:33 +00:00
asapersson@webrtc.org
b60346e951
Reset estimate if no frame has been seen for a certain time (to avoid large jitter if stop sending).
...
Add delay before start processing after a reset.
BUG=1577
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8699006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5561 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-17 19:02:15 +00:00
mallinath@webrtc.org
92fdfebedd
Update talk to 61699344.
...
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5560 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-17 18:49:41 +00:00
mflodman@webrtc.org
e3842897e2
Adding tsan suppression for error introduced in r5555, causing libjingle_unittest to fail on TSan bot.
...
BUG=2931
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8779005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5559 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-17 15:09:39 +00:00
henrik.lundin@webrtc.org
340746aa13
Misc small nits in NetEq
...
Fixing a few small things found recently. This is mostly cosmetics.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8749005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5558 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-17 11:37:16 +00:00
hta@webrtc.org
1009798b31
Demo of multi-pass encode - used for testing limits.
...
This demo creates a sequence of PeerConnections, and passes
a videostream through all of them.
This allows one to test how many PeerConnections and how
many encodes/decodes the implementation will support before
breaking down or slowing down enough to be unusable.
BUG=
R=fischman@webrtc.org , hta@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5557 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-15 06:13:41 +00:00
andrew@webrtc.org
f92aaff104
AudioProcessing is not a Module.
...
Remove Module as the base class of AudioProcessing. The inherited
methods were all no-ops.
TBR=bjornv
Review URL: https://webrtc-codereview.appspot.com/8779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5556 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-15 04:22:49 +00:00
henrike@webrtc.org
b8c254abd6
(Auto)update libjingle 61549749-> 61608469
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5555 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 23:38:45 +00:00
bjornv@webrtc.org
e2fc13e42f
Refactoring common_audio/signal_processing: Removed two macros used by isac only.
...
Removed a macro for malloc() and one for free(). They are only used by the audio codec isac, where I replaced the macro with its implementation.
Further, the includes were updated with full paths and put in alphabetical order.
BUG=N/A
TESTED=trybots,module_tests,module_unittests
R=turaj@webrtc.org , turajs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5554 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 23:12:34 +00:00
fischman@webrtc.org
c5d506a106
AppRTCDemo(android): clarified README on how to launch app using adb.
...
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8689005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5553 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 17:55:13 +00:00
stefan@webrtc.org
505f2a0348
Disabling WebRtcSessionTest.TestIceStatesBundle under memcheck.
...
BUG=2924
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8699005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5552 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 12:38:06 +00:00
stefan@webrtc.org
9075d519a2
Adding a critical section missing in r5543.
...
This fixes a race caught by the linux tsan bot.
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5551 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 09:45:58 +00:00
fischman@webrtc.org
a3708ecdfe
PeerConnectionTest(java): unbreak following 61460797-p10
...
BUG=1414
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5550 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 01:51:33 +00:00
mallinath@webrtc.org
385857dfd4
Update talk to 61549749.
...
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 00:56:12 +00:00
wu@webrtc.org
b9a088b920
Update talk to 61538839.
...
TBR=mallinath
Review URL: https://webrtc-codereview.appspot.com/8669005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5548 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 23:18:49 +00:00
wu@webrtc.org
0de29504ab
Revert 5545 "Update libjingle to 61514460"
...
> Update libjingle to 61514460
>
> TBR=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/8649004
TBR=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5547 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 19:54:28 +00:00
andrew@webrtc.org
38bf249049
Initialize output_will_be_muted_.
...
TBR=aluebs
Review URL: https://webrtc-codereview.appspot.com/8659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5546 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 17:43:44 +00:00
xians@webrtc.org
e749c9ebdb
Update libjingle to 61514460
...
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5545 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 15:09:40 +00:00
asapersson@webrtc.org
8f690bc222
Increase overuse and normal use thresholds for Mac.
...
BUG=1577
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5544 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 14:43:18 +00:00
stefan@webrtc.org
ae2563ae2f
Fixes a race when writing to send_padding_.
...
TEST=trybots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5543 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 13:48:38 +00:00
kjellander@webrtc.org
12cb88cab9
Add check to verify tree is open to PRESUBMIT.py.
...
This will disallow commits when our tree is closed.
BUG=chromium:342743
TEST=ran git cl presubmit with an open tree (no error). Then I closed the tree at http://webrtc-status.appspot.com and ran it again, got this message:
Tree state is: closed
***************
Tree is temporarily closed (testing presubmit hook real quick)
http://webrtc-status.appspot.com/current?format=json
***************
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5542 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 11:53:43 +00:00
henrik.lundin@webrtc.org
fcfc6a990e
Small refactoring of NetEq unittest for CNG with clock drift
...
Converting the test to a method within the test fixture, and setting
up two tests that call this method. One for positive and one for
negative clock drift. The one with positive clock drift is disabled
for now since it does not pass, but will be re-enabled shortly.
This change is only made for NetEq4.
R=tlegrand@google.com
Review URL: https://webrtc-codereview.appspot.com/8599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5541 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 11:42:28 +00:00
fischman@webrtc.org
3eda643a91
PeerConnection(java): added MediaConstraints support to AudioSource, now fed to AudioTrack.
...
BUG=2912
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5540 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 04:01:04 +00:00
fischman@webrtc.org
540acde5b3
PeerConnection(java): use MediaCodec for HW-accelerated video encode where available.
...
Still disabled by default until https://code.google.com/p/webrtc/issues/detail?id=2899 is resolved.
Also (because I needed them during development):
- make AppRTCDemo "debuggable" for extra JNI checks
- honor audio constraints served by apprtc.appspot.com
- don't "restart" video when it hasn't been stopped (affects running with the
screen off)
BUG=2575
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/8269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5539 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 03:56:14 +00:00
andrew@webrtc.org
17342e5092
Add a method to inform AudioProcessing that its output will be muted.
...
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5538 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 22:28:31 +00:00
jiayl@webrtc.org
de782180b0
Change the type of propagation delta from int64 to int.
...
The delta value never exceeds the range of int. Changing it to int will save memory and copying cost.
BUG=2910
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5537 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 19:19:23 +00:00
andrew@webrtc.org
07b5950c12
Initialize key_pressed_.
...
Was resulting in an error on Mac Asan:
[ RUN ] ApmTest.DebugDump
[libprotobuf FATAL ../../third_party/protobuf/src/google/protobuf/message_lite.cc:224] CHECK failed: !coded_out.HadError():
TBR=aluebs
Review URL: https://webrtc-codereview.appspot.com/8539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5536 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 16:41:13 +00:00
andrew@webrtc.org
ce8e077cf0
Add a keypress field to the audioproc debug proto.
...
Log the value in AudioProcessing, and unpack it to a new file in the
unpacking tool.
TESTED=
- The new tool can unpack old dumps.
- The old tool can unpack new dumps (without keypress.bool).
- Unpacking a new dump from voe_cmd_test produces a keypress.bool that
appears correct when examined.
R=aluebs@webrtc.org , bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8509005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5535 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 15:28:30 +00:00
pbos@webrtc.org
8118f1861f
Set pacing bitrates in SetEncoder.
...
Before the change no padding was allowed before the first remote bitrate
estimation was received. This bitrate estimation is based on what's
actually sent. In tests I set codec.startBitrate to 300 instead of
325, which incidentally means that only the first layer gets encoded.
As we only send ~150kbps instead of 300, the first REMB will
significantly pull down the remote bitrate estimate instead of keeping
the existing rate, even though there's no problem with the link.
This was detected in RampUpTest.PacingWithRtx,
(send_config.codec.startBitrate=300), which caused the tests to become a
lot slower, and flake out more. By allowing padding initially we're able
to keep our initial bitrate estimate.
R=stefan@webrtc.org
TEST=Running RampUpTest.WithPacingAndRtx with startBandwidth=300.
BUG=
Review URL: https://webrtc-codereview.appspot.com/8529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5534 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 14:50:29 +00:00
solenberg@webrtc.org
67e70442b5
Remove unused and not working voe_extended_test.
...
BUG=2913
R=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5533 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 09:58:49 +00:00
pbos@webrtc.org
5591046ab1
.gitignore: + /third_party/{clang_format,usrcsctp}
...
clang_format and usrcsctp are both synced in through gclient and should
be suppressed.
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8159004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5532 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 09:33:22 +00:00
jiayl@webrtc.org
14d80793a8
PeerConnectionClient needs to initialize SSL.
...
BUG=2911
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5531 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 00:41:59 +00:00
andrew@webrtc.org
b659e2844d
Reduce mixing threshold in test to avoid flakiness.
...
Flake observed here:
http://chromegw/i/client.webrtc/builders/Win32%20Release%20%5Blarge%20tests%5D/builds/953/steps/voe_auto_test/logs/stdio
TBR=andresp
Review URL: https://webrtc-codereview.appspot.com/8489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5530 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 21:52:50 +00:00
andrew@webrtc.org
75dd2885c5
Add an interface for accepting keypress signals to AudioProcessing.
...
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5529 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 20:52:30 +00:00
andrew@webrtc.org
aa1278de46
Rename merged webrtc lib to libwebrtc_merged.a.
...
The name "libwebrtc.a" was conflicting with the newish webrtc target,
resulting in this error:
$ ./webrtc/build/gyp_webrtc merged_lib.gyp
$ ninja -C out/Debug
ninja: warning: multiple rules generate libwebrtc.a. builds involving
this target will not be correct; continuing anyway
ninja: error: dependency cycle: no_op -> libwebrtc.a -> no_op
BUG=b/12955740
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8409005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5528 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 18:22:29 +00:00
fischman@webrtc.org
8685af7ea0
Remove "Too long processing time of Incoming frame" logspam.
...
This isn't indicative of anything actionable and spams android logcat with times
in the 10-30ms range several times per second.
BUG=2732
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5527 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 17:48:11 +00:00
turaj@webrtc.org
a80be4b23c
Add boundary checking to supress gcc 4.8.3 warning.
...
BUG=2888
Test=try, voe_cmd_test
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5526 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 16:38:45 +00:00
solenberg@webrtc.org
fc320466d1
Remove ViE external encryption API.
...
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8079005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5525 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 15:27:49 +00:00
michaelbai@google.com
82ebb463fd
Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file.
...
This patch removes generate_asm_header.gypi and uses libvpx's obj_int_extract and
unpack_lib_posix to generate offset header files.
It make the simliar feature's implementation consistent.
R=andrew@webrtc.org , fischman@webrtc.org , fischman@chromium.org
BUG=334447
Committed: https://code.google.com/p/webrtc/source/detail?r=5517
Review URL: https://webrtc-codereview.appspot.com/7769006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5524 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 04:48:27 +00:00
wjia@webrtc.org
dd82fa726c
Revert 5516 "Thread annotation of talk_base::CriticalSection."
...
r5516 failed compilation on builds with enable_webrtc=0.
> Thread annotation of talk_base::CriticalSection.
>
> Also enabling -Wthread-safety in talk/build/common.gypi for clang on
> Linux. Thread annotations are compile-time checks that for instance
> certain locks are held before accessing a value.
>
> BUG=
> TEST=Local GUARDED_BY() annotations.
> R=andresp@webrtc.org , fischman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/8189004
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 23:20:15 +00:00
andrew@webrtc.org
16c08f03da
Restore mixing integration tests.
...
These high level tests were disabled over time. Since they depend on
real-time results and the filesystem, they tended to be flaky on the
bots. We now give it a very generous 1 second to start up all channels
before verification and a further relaxed file length check. If we
continue to see problems, I will up the startup delay.
The restored tests would have caught the AGC bug fixed here:
https://code.google.com/p/webrtc/source/detail?r=5454
Add a new "real audio" stress test to exercise more code paths. This
would have caught the refactor bug fixed here:
https://code.google.com/p/webrtc/source/detail?r=5437
BUG=2164,2844
TESTED=git try. Verified it would have caught the aforementioned bugs
by reintroducing them.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5522 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 23:04:39 +00:00
kjellander@webrtc.org
c68d046bb1
Fix BUILD.gn to load all Chromium GN configurations.
...
After troubleshooting with brettw@chromium.org we
found that this line is needed in order to get the
build/config/BUILD.gn configurations loaded.
This should solve the runhooks failures in
http://build.chromium.org/p/client.webrtc.fyi/waterfall
and make it possible to generate projects with
GYP_DEFINES="disable_glibcxx_debug=1"
(which fails today).
TEST=Ran on Linux:
GYP_DEFINES=build_with_tool=tsan gclient runhooks
without and with this patch applied (it fails without).
R=andrew@webrtc.org , brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/8379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5521 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 21:28:55 +00:00
michaelbai@google.com
a65abf9d3a
Revert "Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file."
...
This reverts commit 7686f0ddda
.
BUG=
TBR=andrew@webrtc.org , fischman@webrtc.org ,
Review URL: https://webrtc-codereview.appspot.com/8369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5520 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 19:26:26 +00:00
jiayl@webrtc.org
1f64f06784
Add stats of incoming frame delays for debugging bandwidth estimation.
...
BUG=crbug/338380
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5519 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 19:12:14 +00:00
fischman@webrtc.org
82387e4608
Add ability to receive calls for iOS
...
BUG=2701
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7989005
Patch from Sajid Hussain <shussain@temasys.com.sg>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5518 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 18:47:11 +00:00
michaelbai@google.com
7686f0ddda
Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file.
...
This patch removes generate_asm_header.gypi and uses libvpx's obj_int_extract and
unpack_lib_posix to generate offset header files.
It make the simliar feature's implementation consistent.
R=andrew@webrtc.org , fischman@webrtc.org , fischman@chromium.org
BUG=334447
Review URL: https://webrtc-codereview.appspot.com/7769006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5517 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 17:42:34 +00:00