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Author SHA1 Message Date
henrika@webrtc.org
b7a91fa95a Removes VoERTP_RTCP::InsertExtraRTPPacket. 
Reasons for removing:

- Feels like a complete hack IMHO.
- Not used by any client.
- Unclear functionality regarding time stamp, marker bit etc.
- Causes several issues in tests due to a bad design which mainly depends on the fact that this API "breaks" an ongoing data/packet flow and it complicates the threading model and creates risks for deadlock and memory corruption. Not worth trying to fix given the very unclear benefit of maintaining the API. Better to remove the API instead.
- We also see lots of TSan races and memcheck errors related to this API.

BUG=2296,2240
R=mflodman@webrtc.org, niklas.enbom@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5574 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 08:58:08 +00:00
sergeyu@chromium.org
e384104166 Fix DesktopAndCursorComposer not to crash 
DesktopAndCursorComposer was crashing when screen/window
capturer returns a NULL frame due to an error.

BUG=crbug.com/344093
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5573 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 23:26:34 +00:00
henrike@webrtc.org
5cf3e8f0f0 (Auto)update libjingle $LAST_P10_REVISION-> $NEW_P10_REVISION 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5572 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 22:28:52 +00:00
andrew@webrtc.org
27c6980239 Move the volume quantization workaround from VoE to AGC. 
Voice engine shouldn't really have to manage this. Instead, have AGC
keep track of the last input volume, so that it can avoid getting stuck
under coarsely quantized conditions.

Add a test to verify the behavior.

TESTED=unittests, and observed that AGC didn't get stuck on a MacBook
where this problem can actually occur.

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5571 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 20:24:56 +00:00
solenberg@webrtc.org
00844d7bef Remove obsolete voe_unit_test. 
BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5570 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 18:50:50 +00:00
fischman@webrtc.org
358e3367a3 PeerConnection(java): enable HW encoder on N5 for standalone build. 
Now that bug 2899 is fixed (r5562) packet-loss is recoverable.  Yay.

BUG=2575
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/8869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5568 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 17:29:37 +00:00
fischman@webrtc.org
c2d75e0708 PeerConnection(java): account for thread shutdown vagaries. 
Android's JVM requires threads to detach before they exit, but ONLY if
they needed to AttachCurrentThread.  Conversly, threads that were
attached by the JVM (e.g. the result of making a native call from Java)
must NOT be detached by the application.  This is bug 2441.

The fix for the above is to only pthread_setspecific() for threads that
Attach(), not for already-attached threads.  To ensure that we only
detach Attached threads, added a GetEnv() call to ThreadDestructor(),
which revealed that Oracle's JVM can overly-eagerly clear TLS accounting
data, effectively detaching threads without their consent at shutdown.
Work around this with a specific check.

To guard against (some) regression, added a variant of PeerConnectionTest
that runs on a non-main thread.  This revealed a bug in LinuxDeviceManager
which implicitly assumes its talk_base::Thread has already been
initialized.  Fixed that here too.

BUG=2441
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5567 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 16:57:36 +00:00
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