115 Commits

Author SHA1 Message Date
agalusza@google.com
d818dcb939 Sets up framework for decoding with errors: collects frame sizes (in number of packets) in JB and passes this information to VCMSessionInfo with rtt_ms as FrameData.
R=marpan@google.com, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1841004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4424 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-29 21:48:11 +00:00
tnakamura@webrtc.org
aa4d96a134 Revert r4301
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 19:25:04 +00:00
pbos@webrtc.org
a4407329d4 Include files from webrtc/.. paths in video_coding/.
BUG=1662
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1783006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4348 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 12:32:05 +00:00
hclam@chromium.org
1a7b9b94be Cleanup WebRTC tracing
The goal of this change is to:
1. Remove unused tracing events.
2. Organize tracing events to facilitate measurement of end to end latency.

The major change in this CL is to use ASYNC_STEP such that operation
flow can be traced for the same frame.

R=marpan@webrtc.org, pwestin@webrtc.org, turaj@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1761004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4308 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-08 21:31:18 +00:00
stefan@webrtc.org
66b2e5c05a Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
rtp_rtcp implementation.

This refactoring significantly reduces the receive-side RTP parser and receiver
complexity, and makes it possible to implement RTX correctly by having two
instances of receive-statistics.

With this change the dead-or-alive and packet timeout APIs are removed.

TEST=trybots, vie_auto_test, voe_auto_test
BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1745004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-05 14:30:48 +00:00
pbos@webrtc.org
d900e8bea8 Proper spacing for end-of-namespace comments.
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-03 15:12:26 +00:00
hclam@chromium.org
5616abadf5 Suppress excessive logging in video_coding
Only prints the warning message if a frame was dropped.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1735004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4278 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-27 19:47:40 +00:00
stefan@webrtc.org
4cf1a8af69 Removes kStateFree and kStateDecoding, added a free_frames_ list which simplifies finding a free frame.
The idea is to have all frames not in use be stored in free_frames_, and whenever a packet from a new frame arrives we can just pop a frame from free_frames_. When a frame is grabbed for decoding it will be removed from all lists, and will be added to free_frames_ when it's returned to the jitter buffer.

We should be able to remove the state enum completely later, as their state is defined by the list they are in. But I'll keep it around for now to simplify the cl.

TEST=try bots and vie_auto_test --automated
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1721004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4273 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-27 15:20:14 +00:00
mikhal@webrtc.org
9ca7360b97 VCM: removing max jitter estimate
BUG= 1921
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1690004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4249 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-20 20:13:07 +00:00
hclam@chromium.org
30fb7b83d5 Add a log message to see video delay break down
Shows video delay in terms of:
1. Min playout delay
2. Jitter delay
3. Max decode time
4. Render delay

R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1674004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4242 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-18 21:37:09 +00:00
stefan@webrtc.org
50fb4afade Revert r4145 "Revert 4127 "Switch frame list implementation to std::map.""
TBR=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1678004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4233 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17 07:33:58 +00:00
stefan@webrtc.org
c8b29a2feb Revert r4146 "Revert 4104 "Refactor jitter buffer to use separate lists for de...""
TBR=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1677004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4232 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17 07:13:16 +00:00
kjellander@webrtc.org
63e988856e Merge more tests into modules_{unit,integration}tests.
A new test target named 'modules_integrationtests' is created
and the following test targets were merged into it:
* audio_coding_module_test
* test_fec
* video_coding_integrationtests
* vp8_integrationtests

A couple of other targets were merged into modules_unittests:
* audio_coding_unittests
* audioproc_unittest
* common_unittests
* video_coding_unittests
* video_processing_unittests
* vp8_unittests

I wasn't able to merge audio_decoder_unittests and neteq_unittests due to
conflicts with different defines in these tests.

Some tests that have special requirements aren't merged into
modules_integrationtests yet. I took the opportunity to rename them
since the bot configs will need to be update anyway:
* audio_device_test_api -> audio_device_integrationtests
* video_capture_module_test -> video_capture_integrationtests
* video_render_module_test -> video_render_integrationtests

Exclude files were added for modules_integrationtests to make sure
the memcheck and tsan bots doesn't tests that are too slow
(audio_coding_module_test and vp8_integrationtests were previously
disabled on those bots).

Suppressions for AudioCodingModuleTest needed to be added to get
modules_integrationtests to pass memcheck (even if the test is
excluded from execution).

BUG=1843
TEST=local execution on Linux and trybots (passing except the merged tests of course)
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1656004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4228 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-14 20:09:44 +00:00
fischman@webrtc.org
e001b57d84 Do not hold a lock when calling VCMReceiveCallback::FrameToRender.
DecodedImageCallback is allowed to be called on a thread different from decoding thread. To avoid the deadlock in VCMDecodedFrameCallback::Decoded, VCMDecodedFrameCallback::_critSect  should not be held while calling VCMReceiveCallback::FrameToRender.

BUG=1832
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1570004

Patch from Wu-Cheng Li <wuchengli@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4162 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 03:29:37 +00:00
hclam@chromium.org
b1bba167f4 Prevent excessive logging in jitter buffer
Jitter buffer logs a message when it is going to recycle frames. This adds a
lot of noise even in normal operation. This change make sure only critical
cases are logged.

R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1580007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4150 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-31 18:52:16 +00:00
tnakamura@webrtc.org
694cdc6e84 Revert 4104 "Refactor jitter buffer to use separate lists for de..."
Reason - leading suspect of video frame corruption tracked in http://b/9216252
Note that if this turns out to not be the cause, be sure to re-revert both this change and r4145.

> Refactor jitter buffer to use separate lists for decodable and incomplete frames.
> 
> This changes the design of the jitter buffer to keeping track of decodable frames from the point when packets are inserted in the buffer, instead of searching for decodable frames when they are needed.
> 
> To accomplish this the frame_list_, which previously contained all frames (incomplete or complete, continuous or not), is split into a list of decodable_frames_ (complete, continuous) and a list of incomplete_frames_ (either incomplete or non-continuous). These frame lists are updated every time a packet is inserted.
> 
> This is another step in the direction of doing most of the work in the jitter buffer only once, when packets are inserted, instead of doing it every time we look for a frame or try to get a nack list.
> 
> BUG=1798
> TEST=vie_auto_test, trybots
> R=mikhal@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1522005

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1586007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4146 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-31 16:09:48 +00:00
tnakamura@webrtc.org
4d9c07ad6d Revert 4127 "Switch frame list implementation to std::map."
We want to revert r4104 for b/9216252, but because r4127 was built on top of r4104, we need to revert r4127 first. We'll un/re-revert this if we discover that r4104 is not to blame.


> Switch frame list implementation to std::map.
> 
> This reduces the complexity of insert and find (by timestamp) from linear to logarithmic, which has a big impact on large frame lists.
> 
> BUG=1726
> TEST=trybots, vie_auto_test --automated
> R=mikhal@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1561005

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1590005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4145 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-31 16:06:01 +00:00
mikhal@webrtc.org
adc64a7216 VCM/Timing: Setting clear names to members & methods
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1524004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4140 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-30 16:20:18 +00:00
jiayl@webrtc.org
046bc448d5 Fixes the frameRate stats by grouping the frames by timestamp.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1536004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4138 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29 16:33:46 +00:00
stefan@webrtc.org
ace7ad2302 Switch frame list implementation to std::map.
This reduces the complexity of insert and find (by timestamp) from linear to logarithmic, which has a big impact on large frame lists.

BUG=1726
TEST=trybots, vie_auto_test --automated
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1561005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4127 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29 07:41:48 +00:00
stefan@webrtc.org
7f3f8bc5a6 Refactor jitter buffer to use separate lists for decodable and incomplete frames.
This changes the design of the jitter buffer to keeping track of decodable frames from the point when packets are inserted in the buffer, instead of searching for decodable frames when they are needed.

To accomplish this the frame_list_, which previously contained all frames (incomplete or complete, continuous or not), is split into a list of decodable_frames_ (complete, continuous) and a list of incomplete_frames_ (either incomplete or non-continuous). These frame lists are updated every time a packet is inserted.

This is another step in the direction of doing most of the work in the jitter buffer only once, when packets are inserted, instead of doing it every time we look for a frame or try to get a nack list.

BUG=1798
TEST=vie_auto_test, trybots
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1522005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4104 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-27 07:02:45 +00:00
stefan@webrtc.org
c74c3c2447 Adds integration test for RTX and fixes bugs found.
BUG=1811
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4096 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-23 13:48:22 +00:00
stefan@webrtc.org
a7dc37d568 Log the type of recycled frames.
Also correct the logging of incoming key frame packets.

BUG=1814
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1537004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4090 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-23 07:21:05 +00:00
hclam@chromium.org
8c49c1eab3 Log a message when a key frame packet is received
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1518004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4089 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-22 21:18:59 +00:00
mikhal@webrtc.org
2eaf98b38b Refactor VCM/Timing.
No changes in functionality.

R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/1514004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4081 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-21 17:58:43 +00:00
stefan@webrtc.org
3417eb49f6 Consolidate GetFrame and InsertPacket and move NACK list processing to after a packet has been successfully inserted.
TEST=trybots
BUG=1799
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4080 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-21 15:25:53 +00:00
stefan@webrtc.org
9f557c140e Improve wraparound handling in the render time extrapolator.
This was actually working as intended, but as r3970 changed when render timestamps were extrapolated to when a frame was taken out for decoding, the wraparound could have happened in the Update() step before it had happened in the ExtrapolateLocalTime() step. This causes render timestamps to be generated 13 hours into the future.

TEST=trybots
BUG=1787
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1497004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4055 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-17 12:55:07 +00:00
hclam@chromium.org
fe307e1332 Add one unit test for NACKing a key frame
Adding a test case that wasn't covered. This new test is passing.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1475004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4051 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 21:19:59 +00:00
hclam@chromium.org
b3e5acfb66 Cleanup traces in WebRTC
Remove some unused traces and add a trace counter for encoded video size.

R=holmer@google.com, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1476004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4050 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 21:13:02 +00:00
pbos@webrtc.org
b9bb3d1e7d Avoid resetting encoder on identical settings.
BUG=1681
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1481005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4049 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 18:40:48 +00:00
marpan@webrtc.org
890f6092e6 Bugfix: VCM would report wrong sentBitrate
issue: https://code.google.com/p/webrtc/issues/detail?id=1755

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1484004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4048 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 15:38:44 +00:00
stefan@webrtc.org
2038214c77 Log too long non-decodable duration events.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1488004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4043 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 11:39:06 +00:00
mikhal@webrtc.org
cb20a5b2d7 VCM/JB: Bug fix in ExtractAndSetDecode
BUG=1771
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1466005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4035 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-15 17:10:44 +00:00
stefan@webrtc.org
1673481ed7 Fixes a bug where the render buffer size (and indirectly the non-continuous duration) was computed incorrectly.
BUG=1769
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1473004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4026 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 12:00:47 +00:00
justinlin@chromium.org
7bfb3a3227 Add more tracing for key frames.
R=mallinath@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1428004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4015 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 22:59:00 +00:00
phoglund@webrtc.org
43bf6ce322 Revert 4008 "Avoid resetting video encoder for similar configs."
> Avoid resetting video encoder for similar configs.
> 
> BUG=1681
> R=holmer@google.com, mflodman@webrtc.org, stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1442006

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1431005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4010 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 15:39:26 +00:00
pbos@webrtc.org
aa4efd1535 Avoid resetting video encoder for similar configs.
BUG=1681
R=holmer@google.com, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1442006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4008 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 11:27:16 +00:00
stefan@webrtc.org
d98e784f5f Fix VCMProcessTimer::TimeUntilProcess() unsigned-integer underflow problem.
BUG=1665
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1341004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3979 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-08 06:38:53 +00:00
stefan@webrtc.org
ef14488d03 Trigger a PLI if the duration of non-decodable frames exceeds a threshold.
BUG=1663
R=mikhal@webrtc.org, ronghuawu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3975 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 19:16:33 +00:00
mikhal@webrtc.org
8f86cc8712 VCM/Receiver: Return null when can't extract frame.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1435004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3974 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 18:05:21 +00:00
mikhal@webrtc.org
474e915272 Relanding 3962: VCM/JB: Porting jitter_buffer_test to gtest
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1434004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3971 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 16:55:03 +00:00
mikhal@webrtc.org
759b041019 Relanding r3952: VCM: Updating receiver logic
BUG=r1734
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1433004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3970 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 16:36:00 +00:00
mikhal@webrtc.org
9c7685f9a6 VCM/JB: Break and skip to key if possible
BUG=1734
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1421004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3969 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 16:07:52 +00:00
pbos@webrtc.org
3004c79c6a Fix clang errors in non-GYP_DEFINES=clang=1 build
BUG=1623
R=stefan@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1368004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3968 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 12:36:21 +00:00
stefan@webrtc.org
d3a1959678 Fix jitter buffer unittest.
TBR=mflodman@webrtc.org
BUG=1737

Review URL: https://webrtc-codereview.appspot.com/1430005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3967 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 12:35:58 +00:00
stefan@webrtc.org
a5dee33639 Correctly add packets to nack list when sequence number wraps.
BUG=1737
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1427004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3966 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 11:11:17 +00:00
stefan@webrtc.org
4ce19b1664 Revert r3952 "VCM: Updating receiver logic"
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1410005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3963 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-06 13:16:51 +00:00
stefan@webrtc.org
273759048c Revert r3956 "VCM/JB: Porting jitter_buffer_test to gtest."
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1408005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3962 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-06 13:12:58 +00:00
mikhal@webrtc.org
45f2da0920 VCM/JB: Porting jitter_buffer_test to gtest.
Tests were not modified, but ported as is.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1391004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3956 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-03 22:22:46 +00:00
mikhal@webrtc.org
d3cd565ecf VCM: Updating receiver logic
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1363005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3952 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-03 17:54:18 +00:00