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Author SHA1 Message Date
andrew@webrtc.org
8f69330310 Replace scoped_array<T> with scoped_ptr<T[]>. 
scoped_array is deprecated. This was done using a Chromium clang tool:
http://src.chromium.org/viewvc/chrome/trunk/src/tools/clang/rewrite_scoped_ar...

except for the few not-built-on-Linux files which were updated manually.

TESTED=trybots
BUG=2515
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5985 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 23:10:28 +00:00
buildbot@webrtc.org
658a94595d (Auto)update libjingle 65619249-> 65622932 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5984 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 22:03:30 +00:00
buildbot@webrtc.org
ff90ed6e96 (Auto)update libjingle 65561104-> 65619249 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5983 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 21:12:10 +00:00
andrew@webrtc.org
2eceb8ef46 Roll third_party/opus 258909:262302 
Suppresses the nanny "not-optimized" warning.

TBR=minyue
BUG=2864

Review URL: https://webrtc-codereview.appspot.com/14379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5982 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 15:29:33 +00:00
stefan@webrtc.org
0175d76c72 Fix leak in remote bitrate estimator tests introduced in r5980 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5981 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 11:38:57 +00:00
stefan@webrtc.org
4f616a02bd Support for simulating multiple independent flows in a network. 
R=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5980 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 10:59:24 +00:00
asapersson@webrtc.org
46106f2a05 Casting char to int in logs. 
BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12369006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5979 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 07:02:52 +00:00
buildbot@webrtc.org
2b93402e36 (Auto)update libjingle 65484212-> 65561104 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5978 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 00:18:27 +00:00
jiayl@webrtc.org
cc1ba15fe7 Returns a NULL frame on all platforms if the captured window is closed. 
Part of the fix for crbug/360181.
On Mac/Linux, it previously continues capturing even if the window is closed.
Now it stops by returning a NULL frame.
On Windows, it used to stop capturing when the window is minimized. Now fixed to match other platforms.
Note: the crbug still needs a chrome side fix to close the notification bar.
This fix only stops the stream (i.e. stream onended event fired).

BUG=crbug/360181
TESTED=manually tested in Chrome
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/12329007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5977 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 23:45:56 +00:00
wu@webrtc.org
cd70119a10 Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. 
BUG=3111
TEST=new performance tests
R=niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5976 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 22:10:24 +00:00
wu@webrtc.org
93fd25c20c * Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus. 
* Cast rtp header extension to int in log in rtp_utility.cc.

BUG=3237
TEST=try bots
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5975 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 20:33:08 +00:00
henrik.lundin@webrtc.org
439a4c49f9 Add an output capacity parameter to ACMResampler::Resample10Msec() 
Also adding a unit tests to make sure that a desired output frequency
of 0 passed to AudioCodingModule::PlayoutData10Ms() is invalid.

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5974 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 19:05:33 +00:00
andrew@webrtc.org
103657b484 Add keyboard channel support to AudioBuffer. 
Also use local aliases for AudioBuffers for brevity.

BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5973 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 18:28:56 +00:00
henrik.lundin@webrtc.org
d57b8149c2 Fix the Android compilation (better structure for NetEq test libs) 
This change should make the Android targets compile again. The reason
for the failure was a highly dubious structure in the gypi files. With
this fix, the structure is somewhat cleaner. Still room for improvement.

BUG=3254
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5972 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 13:19:04 +00:00
pbos@webrtc.org
5ca6a5387e Remove TraceCallback use from Call. 
Non-global logging isn't supported, and having a per-call logging
dispatch seems over-eager and adds more complexity than it's worth. The
current implementation is racy on initialization due to missing atomics
support. Besides, logging support should be separate from use of Call.

R=mflodman@webrtc.org
BUG=3250,3157

Review URL: https://webrtc-codereview.appspot.com/12329006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5971 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 11:35:33 +00:00
pbos@webrtc.org
a5c8d2c9b3 Rename Start/Stop in Video{Send,Receive}Streams. 
Rename {Start,Stop}{Sending,Receving} to Start/Stop. StartSending
provides no extra information in the context of a VideoSendStream, as
what it does is to send.

R=mflodman@webrtc.org
BUG=3227

Review URL: https://webrtc-codereview.appspot.com/12329005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5970 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 11:13:21 +00:00
henrik.lundin@webrtc.org
0a2277448e Fixing a bug in ACM2 where the output frame energy was incorrectly set 
The value of AudioFrame::energy_ must be set to -1 in order to have the
energy calculated later on in the AudioConferenceMixer module. This was
not the case in ACM2, where the value was set to 0 instead. This
resulted in bad audio for multi-party calls (5 or more participants).

Implemented a unit test to verify ACM output frame.

BUG=3255
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5969 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 08:11:39 +00:00
andrew@webrtc.org
f26c9e8369 Use unique filenames in AudioProcessingTests for parallelization. 
TBR=bjornv
TESTED="gtest-parallel -w 32 --gtest_filter=*AudioProcessingTests*
out/Debug/modules_unittests" passes.

Review URL: https://webrtc-codereview.appspot.com/14369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 03:46:46 +00:00
buildbot@webrtc.org
3f1aa24078 (Auto)update libjingle 65469804-> 65484212 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5967 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 00:00:12 +00:00
jiayl@webrtc.org
0d915ff603 Fix the return value of DtlsTransportChannelWrapper::SendPacket in the case of invalid RTP packet. 
R=juberti@webrtc.org, mallinath@webrtc.org

BUG=3244

Review URL: https://webrtc-codereview.appspot.com/12299006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5966 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-23 20:46:29 +00:00
bjornv@webrtc.org
e9d3760d5c AEC: Adds a reported_delay_enabled_ flag 
Adds a feature to completely turn on or off buffer handling based on reported delay values. During startup, reported delays are controlled differently through, e.g., WEBRTC_UNTRUSTED_DELAY. By default, the feature is enabled giving the same output as before this change.

TESTED=trybots, modules_unittest
R=aluebs@webrtc.org, andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12349005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5965 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-23 13:20:07 +00:00
henrik.lundin@webrtc.org
26e2b687fc Remove ACM1/ACM2 switching from VoiceEngine tests 
The option to run VoiceEngine tests with both ACM1 and ACM2 was
introduced while the two versions of AudioCoding module where both
in use. Now, ACM1 is being deprecated, and the tests should use the
defualt one (ACM2).

BUG=2996
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5964 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-23 08:39:41 +00:00
andrew@webrtc.org
db144429b2 Exclude the new AudioProcessingTest from some sanitizer bots. 
It takes too long.

TBR=kjellander

Review URL: https://webrtc-codereview.appspot.com/12359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5963 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-23 04:16:24 +00:00
andrew@webrtc.org
46b31b17df Restore sample_rate_hz() until Chromium is updated to not use it. 
TBR=bjornv
TESTED=Chromium builds against webrtc head.
BUG=2894

Review URL: https://webrtc-codereview.appspot.com/12349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5962 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-23 03:33:54 +00:00
buildbot@webrtc.org
504fc89f36 (Auto)update libjingle 65394435-> 65417850 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5961 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-23 03:23:19 +00:00
tkchin@webrtc.org
19b1be159e Provide GetStats method in RTCPeerConnection 
BUG=3144
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12069006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5960 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 21:05:38 +00:00
andrew@webrtc.org
ddbb8a2c24 Support arbitrary input/output rates and downmixing in AudioProcessing. 
Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.

- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.

BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 21:00:04 +00:00
henrik.lundin@webrtc.org
34fe0153b9 Reland "Stop using ACM factory in VoiceEngine" 
This change was originally landed as r5954, but had to be reverted in
r5955 due to bots failing. The failures should be fixed in r5956,
so the original change is now relanded.

BUG=2996
TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5958 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 19:04:34 +00:00
andrew@webrtc.org
d59359af4d Remove 44.1 kHz workaround from the iOS AudioDevice. 
Long, long ago, webrtc didn't support audio at 44.1 kHz. As a result we
treated 44.1 kHz audio as 44 kHz. We now have an arbitrary rate
resampler and have no trouble supporting 44.1 (see 1395 for all the
details). I must have missed updating iOS at the time.

This shouldn't result in a visible change as 16 kHz is selected as the
preferred hardware rate.

BUG=1395
R=fischman@webrtc.org, henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5957 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 18:07:49 +00:00
henrik.lundin@webrtc.org
20c71fd1dc Fix a bug in AcmReceiver::NetworkStatistics 
One of the variables were not copied between the structs.

BUG=2996
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5956 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 10:11:21 +00:00
henrik.lundin@webrtc.org
0c108d0b4d Revert "Stop using ACM factory in VoiceEngine" 
Some of the bots where breaking.

TBR=henrika@webrtc.org
BUG=2996

Review URL: https://webrtc-codereview.appspot.com/12319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5955 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 09:44:00 +00:00
henrik.lundin@webrtc.org
139706ec0b Stop using ACM factory in VoiceEngine 
The factory injection was introduces in order to facilitate switching
between ACM1 and ACM2. Now, ACM1 is being deprecated, and this switching
mechanism is no longer needed.

BUG=2996
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5954 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 08:51:21 +00:00
henrik.lundin@webrtc.org
d144bb6812 Let A/V sync test use default AudioCoding module 
This test used to run with both ACM1 and ACM2, to verify sync with both
versions of the module. ACM1 (and NetEq3) is now being deprecated,
wherefore this test should now use the default one (i.e., ACM2).

BUG=2996
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5953 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 08:36:33 +00:00
henrik.lundin@webrtc.org
0c1444c748 Create ACM2 instance when calling AudioCodingModule::Create 
BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12079005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5952 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 08:18:42 +00:00
kjellander@webrtc.org
0a035c8924 Disable tests in common_video_unittests for Dr Memory. 
These tests takes about 1000 seconds to execute under
Dr Memory Full, causing them to be killed if taking more
than 1200 seconds.
* TestScaler.BiLinearScaleTest
* TestScaler.BoxScaleTest

BUG=3247
TBR=zhaoqin@google.com

Review URL: https://webrtc-codereview.appspot.com/12289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5951 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 08:07:03 +00:00
henrik.lundin@webrtc.org
372ae83228 Reland "Make VoiceEngine choose ACM2 by default"" 
This cl was originally committed as r5923, but was reverted in r5926
due to a blocking bug (issue 3143). The blocking bug was resolved in
r5936.

BUG=2996
TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5950 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 07:21:03 +00:00
bjornv@webrtc.org
5964fe0f86 audio_processing: DestroyHandle() now returns void 
The return value was not used anyhow and there is no proper action to be taken if we would have received an error. Hence, in line with issue441 we should return void upon free.

BUG=441
TESTED=trybots,modules_unittest
R=andrew@webrtc.org, aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5949 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 06:52:28 +00:00
bjornv@webrtc.org
2a796720f8 common_audio: VADFree() now returns void 
* Files in audio_coding are not affected since they never use the return value.
* voice_detection in audio_processing does.
* Updated vad_unittest.cc

BUG=441
TESTED=trybots
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12059005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5948 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 04:45:35 +00:00
fbarchard@google.com
3dfabf928c libyuv r1000 roll for DEPS update to new chromium moving location of gold linker on linux. 
BUG=libyuv:323
TESTED=try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12049006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5947 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-21 22:06:06 +00:00
tkchin@webrtc.org
ec3d8ecdcc Fix typo by renaming RTCSessionDescriptonDelegate -> RTCSessionsDescriptionDelegate 
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5946 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-21 18:47:24 +00:00
andrew@webrtc.org
229e16e254 Add resource audio for audio processing tests. 
This is a prerequisite of:
http://review.webrtc.org/9919004/

TBR=bjornv
BUG=2894

Review URL: https://webrtc-codereview.appspot.com/12219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5945 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-20 03:54:46 +00:00
mallinath@webrtc.org
54fd70046d Remove ASSERT in TransportChannelProxy::SetImplementation, when 
proxy already set to same transport channel impl.

Since session can call SetImplementation multiple times with or without BUNDLE, there are cases when SetImplementation is called with same impl (OnRemoteCandidates/PushdownTransportDescription/SetupMux). Also variables in
cricket::TransportProxy like |connecting_| and |negotiated_| are accessed
both between worker thread and signaling threads (which calls for bigger change
on how session interacts with Transport and TransportChannelProxy). I have a created a separate bug to address later issue.

Also if single thread used as worker and signaling thread, we can end up
calling SetLocalDescription and OnRemoteCandidates in same call sequence, which
will end up calling SetImplementation twice.

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12019007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5944 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-19 01:03:33 +00:00
andrew@webrtc.org
f5a33f145b Resampler modifications in preparation for arbitrary audioproc rates. 
- Templatize PushResampler to support int16 and float.
- Add a helper method to PushSincResampler to compute the algorithmic
delay.

This is a prerequisite of:
http://review.webrtc.org/9919004/

BUG=2894
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5943 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-19 00:32:07 +00:00
sergeyu@chromium.org
3d9ec1fed4 Fix multi-monitor support in the screen capturer for Mac. 
This feature was broken in r5471.

BUG=361919
R=jiayl@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=5937

Review URL: https://webrtc-codereview.appspot.com/12109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5942 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-19 00:25:35 +00:00
buildbot@webrtc.org
8e5ec52e76 (Auto)update libjingle 65152644-> 65219629 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5941 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-19 00:00:31 +00:00
sergeyu@chromium.org
7d055a6e63 Revert r5937 "Fix multi-monitor support in the screen capturer for Mac." 
This would break when rolled in chromium because some code in
chromium depends on the code I changed in that change.

TBR=jiayl@webrtc.org
BUG=361919

Review URL: https://webrtc-codereview.appspot.com/12199005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5940 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-18 23:45:38 +00:00
andrew@webrtc.org
0daa8be9d6 Add Chromium's ScopedVector. 
Trivial changes from the original excepting scoped_vector_unittest.cc,
diff here: https://paste.googleplex.com/6664017300946944

This is a prerequisite for:
http://review.webrtc.org/9919004/

TBR=henrike@webrtc.org
BUG=2894

Review URL: https://webrtc-codereview.appspot.com/12179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5938 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-18 21:20:54 +00:00
sergeyu@chromium.org
be7585b150 Fix multi-monitor support in the screen capturer for Mac. 
This feature was broken in r5471.

BUG=361919
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5937 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-18 18:22:41 +00:00
turaj@webrtc.org
a596a389ea Fix iSAC/48000 issue with ACM2. 
Registeration of iSAC into NetEq is through injecting and external AudioDecoder. This is because iSAC encoder and decoder need to share instances for bandwidth estimator to work. When external decoder is registerred, the sampling rate of the decoder had to be specified. iSAC/48000 decoder has a native sampling rate of 32000 Hz, but it has been registered as 48000 Hz decoder.

This CL fixing this issue by letting NetEq to obtain sampling rate of an external coder according to its existing database.

BUG=3143
TEST=voe_cmd_test,modules_unittest,try-bots
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5936 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-17 23:30:49 +00:00
fischman@webrtc.org
29540b1879 Revert "PeerConnectionFactory: delay deletion of owned threads." 
This reverts r5933 because it broke
http://build.chromium.org/p/client.webrtc/builders/Win64%20Release/builds/1598

BUG=3100

Review URL: https://webrtc-codereview.appspot.com/12159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5935 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-17 22:54:30 +00:00