cf1b51b6fb
Moves the display reconfiguration callback into a separate class,
...
so that it can be shared with the cursor monitor when single monitor capturing
is added (https://webrtc-codereview.appspot.com/4679005/ ).
This Cl should have no functionality change.
BUG=2253
R=henrike@webrtc.org , sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/7599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5461 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 21:59:12 +00:00
07e5196414
Added new capture callback interface to pass the capture callback to a specific voe channel from libjingle webrtcvoiceengine.cc.
...
The callback has to go through VoEBaseImpl since VoEChannel is internal to voice engine.
TEST=compile
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7769005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5458 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 13:54:02 +00:00
094ac39b5a
Fix race when deleting video receive streams in Call.
...
BUG=
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5457 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 11:21:58 +00:00
f7c6e743b3
Fix deadlock in video_receiver.cc.
...
In webrtc::vcm::VideoReceiver::ResetDecoder(), the lock order is:
1. take _receiveCritSect,
2. take process_crit_sect_
This conflicts with the follow code path:
1. webrtc::vcm::VideoReceiver::Process(), take process_crit_sect_
call -> webrtc::vcm::VideoReceiver::NackList(),
2. with nackStats=kNackKeyFrameRequest, take _receiveCritSect
BUG=2861
TEST=trybots
R=sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5456 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 10:27:51 +00:00
41907748cb
Connect webrtc::Config to WrappingBitrateEstimator
...
This is the second CL for this change. Connection to the ViE API
remains to be done.
BUG=2698
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5455 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 08:47:15 +00:00
c7c7a531f3
Add Config struct for experimental AGC.
...
Disable in the audio mixer.
TBR=aluebs,bjornv
BUG=2844
Review URL: https://webrtc-codereview.appspot.com/7769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5454 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 04:57:25 +00:00
7433a088d2
Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..."
...
We reverted the r5421 to allow us roll webrtc to chrome without any modifications
to libjingle. Since webrtc is rolled with r5444, we can add back the original CL
and changes to libjingle will be upstreamed in the next roll.
TBR=andresp@webrtc.org
> Revert 5421 "Fix deadlock on register/unregister observer while ..."
>
> Failure to compile on Chromium Internal bots, because of API changes.
>
> http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Mac/builds/2805/steps/compile/logs/stdio
>
> You need to follow the steps mentioned in
> https://docs.google.com/a/google.com/document/d/1aHrmXECnu3-Jovc2-zYI267EaQCYz-IclYyBp9iA9Fc/edit that of a API changer.
>
> Since I will be rolling the libjingle this week, I can push your changes along with libjingle roll, if you prepare the CLs
> as mentioned in the doc.
>
> > Fix deadlock on register/unregister observer while there is a an going callback.
> >
> > BUG=2835
> > R=mallinath@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/7119005
>
> TBR=andresp@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/7679004
TBR=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7729005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5453 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 00:56:02 +00:00
84eb0e952e
Add clean test to NetEq perf test
...
Add another test to NetEqPerformanceTest with no packet losses or
clock drift. The purpose of this test would be to focus on the
"clean" code path, i.e., the path taken when there are no network
problems. The reason is that this code path is presumably much
lighter in complexity, and regressions could easily drown in the
heavier code involved when combating losses and drift.
BUG=2859
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7689005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5452 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-28 21:50:35 +00:00
932b0193e7
VideoCaptureAndroid: stop preview in opposite order of starting.
...
While the SDK documentation doesn't prescribe a required shutdown order, good
hygiene suggests stopping should happen in reverse order of starting. It also
seems to relieve a crash in the system capturer on at least the Galaxy Note 10.
BUG=2793
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5445 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 22:32:05 +00:00
18586d38bc
Revert 5421 "Fix deadlock on register/unregister observer while ..."
...
Failure to compile on Chromium Internal bots, because of API changes.
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Mac/builds/2805/steps/compile/logs/stdio
You need to follow the steps mentioned in
https://docs.google.com/a/google.com/document/d/1aHrmXECnu3-Jovc2-zYI267EaQCYz-IclYyBp9iA9Fc/edit that of a API changer.
Since I will be rolling the libjingle this week, I can push your changes along with libjingle roll, if you prepare the CLs
as mentioned in the doc.
> Fix deadlock on register/unregister observer while there is a an going callback.
>
> BUG=2835
> R=mallinath@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/7119005
TBR=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5444 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 22:00:57 +00:00
a45cac0fb7
Avoid potential dead lock in StreamStatisticianImpl
...
Extract callbacks for rtp/rtcp data, from StreamStatisticianImpl to
ReceiveStatisticsImpl, into separate methods with guards agains having
incorrect lock order.
BUG=2856
R=andresp@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5441 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 16:22:08 +00:00
5314e85926
Race condition in RTPSender::UpdateRtpStats
...
The ssrc should not be access directly from the ssrc_ field, without
holding the send_critsect_ lock. A better way is to just use the SSRC()
getter method.
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7539006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5439 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 13:20:36 +00:00
d9b9560ee5
Drop early packets when not sending in TransportAdapter.
...
Particularly, suppress periodic RTCP packets before
VideoSendStream.StartSending() or VideoReceiveStream.StartReceiving() have been called, respectively.
RTCP packets are sent periodically, by the Process thread, for every ViE channel even those not sending.
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5438 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 13:03:02 +00:00
2397a17c6b
Fix bug introduced during replace of list wrapper with std equivalents in r5378.
...
R=henrika@webrtc.org , pbos@webrtc.org , henrike@webrtc.org
TBR=henrike@webrtc.org
BUG=2164
Review URL: https://webrtc-codereview.appspot.com/7639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5437 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 12:33:30 +00:00
c00adbed73
Race in StreamStatisticianImpl::GetStatistics vs. ::IncomingPacket
...
StreamStatisticianImpl.ssrc_ is protected by stream_lock_, should
be cached while holding lock to avoid race condition.
Also, rtp_callback_ do not need to be called in GetStatistics() at all
BUG=2853
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5435 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 10:42:48 +00:00
99eab02fb1
Fix "field '_testNo' is uninitialized" warnings.
...
BUG=2849
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5434 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 09:30:35 +00:00
c98882dcd3
Always initialize Trace in Call TraceDispatcher.
...
Prevents violation of lock order occuring previously when
RegisterCallback called SetTraceCallback while holding its lock, which
called Print back (which acquires the lock).
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5433 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 09:11:10 +00:00
e84978f3d8
Add a Config parameter to AudioProcessing::Create().
...
Also add a parameter-less version; the (int) version is deprecated and
should be removed.
TBR=aluebs,bjornv
BUG=2844
Review URL: https://webrtc-codereview.appspot.com/7609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5431 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-25 02:09:06 +00:00
57f6c10d00
Android, WebRTCDemo: fixes crash issue when pressing switch camera button on devices with only one camera.
...
BUG=2807(second issue)
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5429 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-24 18:42:12 +00:00
871d949299
Remove loopback setup in RtpRtcpImplTest. Changed to use two separate rtp/rtcp modules.
...
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5426 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-24 13:23:49 +00:00
fd0f267bb1
Add new API (webrtc.gyp:webrtc) to merge_libs.gyp.
...
Required to be able to link new API code against the merged target.
Replaces old dependency on video_engine_core as the new-API target
depends on it for now, and video_engine_core is being phased out.
R=mflodman@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/7519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5424 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-24 10:43:47 +00:00
99a8c7e039
Add trace-based delivery filter to BWE test framework.
...
R=mflodman@webrtc.org , solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5889005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5423 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-24 10:00:27 +00:00
c279a5d72c
Wire up RTX in VideoReceiveStream.
...
Also adds a test to make sure that a retransmitted frame is actually
received and decoded on the remote side. The previous NACK test checked
retransmission, but not that the receiver actually takes care of the
retransmitted packet.
BUG=2399
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5422 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-24 09:30:53 +00:00
8d375c95b7
Fix deadlock on register/unregister observer while there is a an going callback.
...
BUG=2835
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7119005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5421 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-23 23:09:25 +00:00
754de528b7
Fix array declarations in aec_rdft.h.
...
Was causing warnings in Chromium such as:
warning C4742: 'rdft_wk2i' has different alignment in
'webrtc\modules\audio_processing\aec\aec_rdft_sse2.c' and
'webrtc\modules\audio_processing\aec\aec_rdft.c': 4 and 16
BUG=chromium:336620
R=cduvivier@google.com
Review URL: https://webrtc-codereview.appspot.com/7489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5419 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-23 20:55:14 +00:00
e7223e7795
Set NACKed packet to -1 in TestNackRetransmission.
...
Zero is a valid sequence number which may occur even if there are no
retransmissions, this caused the test to flake as an incoming packet
would be mistaken for a retransmission.
BUG=2830
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7509005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5417 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-23 16:14:34 +00:00
0e93257cee
Add callbacks for receive channel RTP statistics
...
This allows a listener to receive new statistics (byte/packet counts, etc) as it
is generated - avoiding the need to poll. This also makes handling stats from
multiple RTP streams more tractable. The change is primarily targeted at the new
video engine API.
TEST=Unit test in ReceiveStatisticsTest.
Integration tests to follow as call tests when fully wired up.
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5416 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-23 10:00:39 +00:00
91db93d24f
Android, fixes crash on devices with only front cameras.
...
BUG=2807
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5415 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-22 21:31:24 +00:00
7de3bb9df9
Output logs to stderr from voe_cmd_test by default.
...
Add a flag --log_file which produces the existing behaviour of dumping
logs of all severities to a file. By default, warnings and errors will
now be output to stderr. This is generally more useful for the testing
done with voe_cmd_test.
TESTED=logs output to stderr by default and to the usual file when the
flag is specified.
R=tnakamura@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6849005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5409 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-21 22:17:43 +00:00
28da47c52f
Android example apps: fixes issue where useful failure information was suppressed.
...
BUG=2808
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5408 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-21 19:03:51 +00:00
7dba27c740
Potential dead lock in receive statistics
...
A dead lock could occur if the following to code paths are called
concurrently:
ReceiveStatisticsImpl::IncomingPacket() ->
StreamStatisticianImpl::IncomingPacket()
StreamStatisticianImpl::GetStatistics() ->
ReceiveStatisticsImpl::StatisticsUpdated()
Solution is to release ReceiveStatisticsImpl lock after lookup/lazy-init of StreamStatisticianImpl. Don't need to hold it when doing the call to StreamStatisticianImpl::IncomingPacket().
BUG=2818
R=asapersson@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5406 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-21 16:33:37 +00:00
32c3247418
Fix for libtalkmobile build error
...
bug=b/12549061
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5404 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-21 16:16:58 +00:00
7ef7df57d8
Removes script for generating supplement.gypi also adds git ignore for tools/gn.
...
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5403 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-21 15:54:56 +00:00
e02d47515f
Set up receiver RTX config using a std::map.
...
This change removes video_payload_type from RtxConfig as it can be
inferred from the map key or config otherwise. Wiring up this config is
part of issue 2399.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5402 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-20 14:43:55 +00:00
efaeda0c76
Add configuration and test for extended RTCP reference time reports to new video api.
...
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5401 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-20 08:34:49 +00:00
32c26eb90b
Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback
...
BUG=N/A
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5400 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-17 23:12:51 +00:00
4985927d36
Implement screen enumeration and individual screen capturing for Windows.
...
BUG=2787
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/7239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5399 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-17 17:19:16 +00:00
ead202b973
Android, OpenSlDemo: fixes issue where app would crash as soon as the application is started.
...
BUG=2801
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7259005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5398 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 23:26:37 +00:00
0af1ffa84d
Android, WebRTCDemo: fix issue where changing remote IP was not working properly.
...
BUG=2783
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5396 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 16:45:15 +00:00
4ffd9c7423
Add full path to headers
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5395 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 09:01:39 +00:00
6a94734d4d
Adds back set_sample_rate_hz() when Init is called in recordings.
...
Recordings that had a AnalyzeReverseStream() call prior to ProcessStream() where aborted due to sample rates being set upon call by ProcessStream(). That change was done in r5346.
Before we have a smarter handling on how to set sample rate automatically, this CL adds back that setting.
BUG=
TESTED=trybots, modules_unittests
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5394 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 08:41:09 +00:00
ea9392d5eb
MIPS optimizations for NS audio processing module
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4139006
Patch from Ljubomir Papuga <lpapuga@mips.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5393 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 07:22:01 +00:00
fb4e256d49
Fix crash in MouseCursor::CopyOf()
...
This issue was causing test failures with the latest webrtc roll.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7249005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5392 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 04:45:35 +00:00
8f35afab8c
Exclude protoc objects from merge_libs.py.
...
BUG=b/12567343
R=wjia@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5391 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 00:31:57 +00:00
7a2ca7c621
Update needed to MockScreenCapturer after new methods addition to webrtc::ScreenCapturer.
...
This change is also must for rolling webrtc in chrome.
R=jiayl@webrtc.org
TBR=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/7219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5389 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-15 19:00:13 +00:00
017b619010
Extends the ScreenCapturer interface for individual display screen cast.
...
Real implementations for each platform will be added in future CLs.
BUG=2787
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/6819005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5386 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-14 18:26:37 +00:00
03cfde2d10
Roll Chromium 238260 -> 243863
...
R=andrew@webrtc.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5385 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-14 17:48:34 +00:00
39fcfd78ae
Remove empty VideoCodecGeneric struct.
...
Struct was added prematurely and triggers a warning with
-Wextern-c-compat in latest clang.
R=henrika@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/7119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5383 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-14 12:55:59 +00:00
d9faa46d57
Changing to using factory methods for some classes in NetEq
...
In this CL, the Expand, Accelerate and PreemptiveExpand objects are
created using factory methods. The factory methods are injected into
NetEqImpl on creation. This is a step towards implementing a no-decode
operation.
BUG=2776
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6999005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5382 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-14 10:18:45 +00:00
4371d4650a
Temporarily disabling some more audio processing tests.
...
R=andrew@webrtc.org , bjornv@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7019005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5380 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-14 08:57:22 +00:00
