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Author SHA1 Message Date
henrik.lundin@webrtc.org
6f6ba6edee Fix issues with sequence number wrap-around in jitter statistics 
Wrap-arounds in sequence numbers (and in timestamps) were not always
treated correctly. This is fixed by introducing two helper functions
for correct comparisons, and by casting to the right word size.

Also added a new member variable to AutomodeInst_t. The new member keeps
track of when the first packet has been registered in the automode code.
This was previously done implicitly (and not very good) using the fact
that the lastSeqNo and lastTimestamp members were initialized to zero.

Two new unit tests were added as part of this CL.
NetEqDecodingTest.SequenceNumberWrap was failing before the fixes were
made; now it is ok.

BUG=2654
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5150 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 13:17:29 +00:00
pbos@webrtc.org
b3cc78de28 Add -Wnon-virtual-dtor warning for C++ code. 
BUG=2659
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4119006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5149 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 11:42:02 +00:00
sprang@webrtc.org
72964bd4fb Make interface destructor virtual 
In summary, do this:

-  ~FrameCountObserver() {}
+  virtual ~FrameCountObserver() {}

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4099005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5148 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 09:09:54 +00:00
asapersson@webrtc.org
8d02f5dc71 Added API for enabling/disabling RTCP Receiver Reference Time extension. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3419005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5147 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 08:57:04 +00:00
asapersson@webrtc.org
54a05518e2 Increase run-time for full stack test for the rtt to be added reliably to the delay measurement. 
BUG=2592
R=holmer@google.com, phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5146 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 07:45:08 +00:00
braveyao@webrtc.org
425e1d0fb9 Typo in vie_autotest_win.cc 
BUG=2637
TEST=AutoTest
R=mflodman@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5145 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 02:17:01 +00:00
henrike@webrtc.org
a750044396 Fixes a crash in VoE when unregistering JNI hooks. 
BUG=11695087
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5144 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 22:32:12 +00:00
wu@webrtc.org
364f204d16 Update talk to 56698267. 
TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/4119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5143 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 21:49:41 +00:00
sprang@webrtc.org
dc50aaeaa8 Interface changes to old api, for use by new api transition. 
BUG=2589
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5142 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 16:47:07 +00:00
asapersson@webrtc.org
b24d33565c Added ViE API for getting overuse measure. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3129005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5141 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 13:51:40 +00:00
pbos@webrtc.org
d29d4e9c08 Deliver I420VideoFrames from VideoRender module. 
Performance issue and simplicity, this implementation skips conversion
to VideoEngine's frame format and then back again to I420VideoFrame.

BUG=2526
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5140 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 13:19:54 +00:00
asapersson@webrtc.org
1ae1d0c471 Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module). 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2383004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5139 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 12:46:11 +00:00
pbos@webrtc.org
27326b6a42 Rename newapi::Transport::SendRTP()->SendRtp(). 
Also fit rampup_tests.cc to use internal::TransportAdapter instead of
implementing its own.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5138 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 12:17:04 +00:00
pbos@webrtc.org
ce90eff345 Rename RTP-extension constants. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5137 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 11:48:56 +00:00
pbos@webrtc.org
53c8573525 Rename video streams' start/stop methods. 
{Start,Stop}{Send,Receive}() -> {Start,Stop}{Sending,Receiving}().

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5136 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 11:36:47 +00:00
pbos@webrtc.org
5a63655ab0 Rename Call::Create{Receive,Send}Stream(). 
Renaming the methods to include Video. Long-term there will hopefully be
AudioSendStream/AudioReceiveStreams as well.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 10:40:25 +00:00
aluebs@webrtc.org
0b72f5863b Add experimental noise suppression dummy API. 
Add this flag to the voe_cmd_test.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5134 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-19 15:17:51 +00:00
sergeyu@chromium.org
5d85819dd2 Fix DesktopAndCursorComposer to restore frames to the original state. 
Screen capturers may reuse frame buffers and they expect that the
frame content isn't changed by the frame consumer.
DesktopAndCursorComposer draws mouse cursor on generated frames and
it was releasing the frames with the mouse cursor on them. Fixed
it to restore frame content erasing mouse cursor before returning
desktop frames.

BUG=crbug.com/316297
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/3899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5133 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-19 02:15:47 +00:00
turaj@webrtc.org
7a05ae5c69 Adding back main() to the test. Now it is possible to choose between ACM1 and ACM2, furthermore, the test can simulate a channel with packet loss and FEC can be activated. Packet loss pattern is based on channel implementation in Channel{.cc,.h}, which currently is a determenistic pattern with 1 every 3rd packet is discarded. 
The main() was deleted in r4731.

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

Review URL: https://webrtc-codereview.appspot.com/2370004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5132 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-18 18:16:53 +00:00
pbos@webrtc.org
9c5fb76662 Exclude AV-sync test from Valgrind platforms. 
Test is performance-dependent and was observed to never sync on the
linux_memcheck bot.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5131 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-18 16:22:50 +00:00
henrik.lundin@webrtc.org
ce8e0936d9 Rename AutoMute to SuspendBelowMinBitrate 
Changes all instances throughout the WebRTC stack.

BUG=2436
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-18 12:18:43 +00:00
stefan@webrtc.org
28bf50f0ec Fix test broken with r5128. 
TBR=pbos@webrtc.org
BUG=2530

Review URL: https://webrtc-codereview.appspot.com/3979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5129 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-18 11:58:24 +00:00
stefan@webrtc.org
b082ade3db Hook up audio/video sync to Call. 
Adds an end-to-end audio/video sync test.

BUG=2530, 2608
TEST=trybots
R=henrika@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5128 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-18 11:45:11 +00:00
stefan@webrtc.org
4cfa6050f6 Fix breakage after introducing new test. 
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3899005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5127 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-15 13:15:56 +00:00
stefan@webrtc.org
69969e2e2f Improve Call tests for RTX. 
Also does some refactoring to reuse RtpRtcpObserver.

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

Review URL: https://webrtc-codereview.appspot.com/3809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5126 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-15 12:32:15 +00:00
henrik.lundin@webrtc.org
6e95d7afab Increment RTP timestamps for padding packets 
This CL makes the padding packets get their own RTP timestamps,
rather than having the same timestamp as the last sent video
packet. The purpose is to solve Issue 2611, where the overuse-
detector does not react to padding packets.

A test was implemented to verify that the padding packets do
get their own timestamps.

BUG=2611
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5125 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-15 08:59:19 +00:00
pbos@webrtc.org
6488761f2e Implement VideoSendStream::SetCodec(). 
Removing assertion that SSRC count should be the same as the number of
streams in the codec. It makes sense that you don't always use the same
number of streams under one call. Dropping resolution due to CPU overuse
for instance can require less streams, but the SSRCs should stay
allocated so that operations can resume when not overusing any more.

This change also means we can get rid of the ugly SendStreamState whose
content wasn't defined. Instead we use SetCodec to change resolution
etc. on the fly. Should something else have to be replaced on the fly
then that functionality simply has to be implemented.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3499005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5123 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-14 08:58:14 +00:00
sergeyu@chromium.org
183c727bca Disable datachannel_unittest.cc 
the test fails to compile because it uses incorrect gmock path (as 
some other tests).

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5121 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-13 22:59:20 +00:00
sergeyu@chromium.org
a23f0ca4ba Update talk to 56619788 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3839005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5120 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-13 22:48:52 +00:00
kjellander@webrtc.org
e8722856f9 Disable all vie_auto_tests on Linux for now (take 2) 
Turns out OS_LINUX is not working in this context
(see http://review.webrtc.org/3539005/)
WEBRTC_LINUX is the right define to use.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5119 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-13 15:51:49 +00:00
kjellander@webrtc.org
c8489852ec Disable all automated vie_auto_tests on Linux for now 
Since the switch from icewm to openbox window manager on
Linux in Chrome infra, causes the test to hang when
creating Windows.

TEST=trybots compile step
BUG=chromium:318760
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3539005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5118 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-13 15:44:54 +00:00
stefan@webrtc.org
9b82f5a6ed Fix for RTX in combination with pacing. 
Retransmissions didn't get sent over RTX when pacing was enabled since
the pacer didn't keep track of whether a packet was a retransmit or not.

BUG=1811
TEST=trybots
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5117 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-13 15:29:21 +00:00
turaj@webrtc.org
03f33709f8 Inject config when creating channels to override the existing one. 
BUG=
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5116 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-13 00:02:48 +00:00
henrik.lundin@webrtc.org
e8433eb115 Reimplementing NetEq4's AudioVector 
The current implementation using std::vector is too slow.
This CL introduces a new implementation, using a regular
array as data container.

In AudioMultiVector::ReadInterleavedFromIndex, a special case for
1 channel was implemented, to further reduce runtime. Finally,
AudioMultiVector::Channels was reimplemented.

The changes in this CL reduces the runtime of neteq4_speed_test
by 33%.

BUG=1363
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5115 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-12 13:15:02 +00:00
asapersson@webrtc.org
38599510df Parse next RTCP XR report block after an unsupported block type. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5114 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-12 08:08:26 +00:00
minyue@webrtc.org
3e427263ee Reducing opus_test runtime to pass Android test 
BUG=2609
R=solenberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5111 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-11 22:03:52 +00:00
andrew@webrtc.org
e03cafaebc MIPS optimizations for AECM audio processing module 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2279005

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5110 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-11 20:10:01 +00:00
andrew@webrtc.org
b0730108a2 Move audio_processing dependencies to a variable. 
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5108 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-11 17:20:27 +00:00
pbos@webrtc.org
57eb858698 Remove ".." from include_dirs in build/common. 
BUG=1662
TEST=compile on trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2332004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-11 10:20:27 +00:00
andrew@webrtc.org
6e908b3adf Remove unnecessary include_dirs from audio_processing. 
TBR=aluebs
TESTED=trybots

Review URL: https://webrtc-codereview.appspot.com/3659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5106 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-08 19:52:05 +00:00
marpan@webrtc.org
00ed170795 Roll libvpx 225010:232686. 
R=andrew@webrtc.org
TBR=ajm@google.com

Review URL: https://webrtc-codereview.appspot.com/3649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5105 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-08 18:37:18 +00:00
andrew@webrtc.org
5973f3a24a Remove unneeded includes from trace_posix.cc. 
TESTED=trybots
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5103 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-08 17:30:07 +00:00
stefan@webrtc.org
48df38114d Fix for making sure that the packet in order checks are done prior to updating the last received packet state. 
Without this fix all packets are considered out-of-order by the rtp receiver, causing the last received state
in the rtp receiver to never get valid.

Also makes sure that only valid timestamps and receive times are used for audio/video sync.

BUG=2608
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-08 15:18:52 +00:00
henrikg@webrtc.org
bff9620116 Fix log build error for Chromium builds. 
This only happens when building in Chromium. Can't roll due to this.

../../third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc: In function 'Window {anonymous}::GetTopLevelWindow(Display*, Window)':
../../third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc:39:7: error: 'LS_INFO' was not declared in this scope
../../third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc:39:7: note: suggested alternative:
../../third_party/webrtc/system_wrappers/interface/logging.h:71:29: note:   'webrtc::LS_INFO'

See for example http://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20%5Blatest%20WebRTC%2Blibjingle%5D/builds/3039/steps/compile/logs/stdio

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5100 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-08 10:37:27 +00:00
kjellander@webrtc.org
4c828e145e Remove update_resources.py as it's no longer used. 
After http://review.webrtc.org/2095004/ has been landed
for normal WebRTC builds, and https://codereview.chromium.org/62273004/
and https://codereview.chromium.org/60513012/ for our Android
APK builds with a Chromium checkout, we should be fine to remove
this script.

I have verified that the runhooks step on the Android testers
is using the download_from_google_storage.py script to pull
the resources from Google Storage.

BUG=webrtc:2294
TEST=a few trybots passing compile step.
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5099 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-08 09:08:36 +00:00
andrew@webrtc.org
f1a48174d4 Replace disabled logging with a restricted logging mode. 
This will enable some low-level webrtc logging in a Chromium build,
while limiting the binary size impact.

For a Mac Release build, it results in an increase to Chrome.app of 37k
and libpeerconnection.so of 25k. For comparison, enabling full logs
costs 230k and 218k respectively.

BUG=b/11470432
TESTED=voe_cmd_test produces logs of the appropriate severity.
R=fischman@webrtc.org, henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5097 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-07 23:47:26 +00:00
elham@webrtc.org
5adc89747a Updated WebRTC version to 3.46 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5093 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-06 22:27:51 +00:00
fbarchard@google.com
a7855a88b3 Fix for xgetbv on Visual Studio 2010. 
BUG=none
TEST=local build of webrtc with 2010.  python build\gyp_chromium --depth=. -G msvs_version=2010 -fninja all.gyp & ninja -C out\Debug
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5092 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-06 21:18:48 +00:00
marpan@webrtc.org
bde3056567 Fix for video_processor_intergration_tests to run in parallel. 
BUG=2601.
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5091 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-06 20:59:29 +00:00
kjellander@webrtc.org
c4225b63bb Update getUserMedia W3C conformance tests. 
This CL updates these tests to the spec as of
http://dev.w3.org/2011/webrtc/editor/archives/20130824/getusermedia.html

There are still a lot of functionality that lacks testing. I've put a bunch of TODOs in there but I'm unlikely to get time to implement them all any time soon...

TEST=local testing with Chrome Canary.
BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5090 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-06 13:26:34 +00:00