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
buildbot@webrtc.org
1a87f529a2
(Auto)update libjingle 65151416-> 65151642
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5934 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 22:41:30 +00:00
fischman@webrtc.org
cea024d672
PeerConnectionFactory: delay deletion of owned threads.
...
Since PeerConnection holds a ref to its creating PeerConnectionFactory, it's
possible for ~PeerConnectionFactory() to be run on its signaling thread.
Deleting a thread from within that thread is sad times, so don't do it.
It would be nicer to avoid having PeerConnection hold a ref to the factory,
and instead require the user to keep the factory alive. Unfortunately that
changes the contract on PeerConnection{,Factory} and it's unclear how to vet
existing callers for safety.
BUG=3100
R=juberti@webrtc.org , noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/11289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5933 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 22:36:00 +00:00
marpan@webrtc.org
b476d36120
Roll libvpx 259973:264320
...
TBR=ajm@google.com
Review URL: https://webrtc-codereview.appspot.com/12069007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5932 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 20:35:03 +00:00
henrike@webrtc.org
aeb0c28193
Update PRESUBMIT.py's list of "DO_NOT_SUBMIT_FILES".
...
BUG=N/A
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12029006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5931 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 14:15:43 +00:00
kwiberg@webrtc.org
e57ae02327
WebRtcAecm_Process: Reduce code duplication
...
BUG=
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5930 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 12:28:33 +00:00
kwiberg@webrtc.org
d2f366f28c
StereoToMono: Remove useless call to WebRtcSpl_SatW32ToW16
...
The max value is ((2**15 - 1) + (2**15 - 1)) >> 1
== (2**16 - 2) >> 1
== 2**15 - 1
which doesn't overflow.
The min value is (-2**15 + -2**15) >> 1
== -2**16 >> 1
== -2**15
which doesn't overflow.
Since those two bracket all possible results, the call to
WebRtcSpl_SatW32ToW16 is redundant.
BUG=
R=andrew@webrtc.org , bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5929 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 12:17:39 +00:00
henrika@webrtc.org
66803489f9
Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs.
...
BUG=3206
R=henrik.lundin@webrtc.org , juberti@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12019005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5928 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 10:45:01 +00:00
henrika@webrtc.org
0f7375504a
Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs.
...
BUG=3206
R=juberti@webrtc.org , niklas.enbom@webrtc.org , tina.legrand@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5927 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 10:38:08 +00:00
henrik.lundin@webrtc.org
e2e9abb3bc
Revert "Make VoiceEngine choose ACM2 by default"
...
The reason for reverting is that Issue 3143 should be resolved
first.
TBR=henrika@webrtc.org
BUG=3143
Review URL: https://webrtc-codereview.appspot.com/12119005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5926 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 10:12:27 +00:00
buildbot@webrtc.org
0b3c6c3191
(Auto)update libjingle 65086785-> 65104022
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5925 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 10:03:57 +00:00
henrik.lundin@webrtc.org
adaf809612
Removing AudioCoding duplicate tests
...
Reverting to using one version of ACM in ACM tests.
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5924 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 08:29:10 +00:00
henrik.lundin@webrtc.org
6cec07f6a7
Make VoiceEngine choose ACM2 by default
...
The use of a factory for ACM will be removed in later CLs.
BUG=2996
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5923 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 08:07:18 +00:00
fischman@webrtc.org
c0a15b7ddc
Fix crashes due to dangling external decoder pointer.
...
When checking whether we need to release external decoder,
we have to do pointer comparison. We can't rely on payload
types, because payload types can be stale (e.g. before we
decode the first video frame after RegisterReceiveCodec).
This leaves a dangling pointer to external decoder, which
leads to crashes later, after we actually delete the
external decoder object.
This change has been verified in Chromecast code tree.
BUG=chromium:335539
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12049004
Patch from Sergey Volk <servolk@chromium.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5922 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 01:22:48 +00:00
buildbot@webrtc.org
39b868bad3
(Auto)update libjingle 65055925-> 65086785
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@5921 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 00:04:39 +00:00
jiayl@webrtc.org
8f88f20af2
Expand the test max wait time from 1000ms to 2000ms.
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The createOffer/createAnswer methods sometimes times out due to slow identity generation under memcheck.
BUG=2838
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5920 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 17:14:21 +00:00
kjellander@webrtc.org
c18729165a
Set include_internal_video_capture=1 for video_capture_tests
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Having this override in the .gypi file avoids having to set it for the bots, which I think is best if we can avoid.
This CL also reverts r5869 so video_capture_tests are compiled for Android again.
BUG=2974,3152
TEST=Successfully ran:
git try -t compile
git try --bot=win_baremetal,mac_baremetal,linux_baremetal -t video_capture_tests
git try --bot=android_apk,android_apk_rel
Verified the change in webrtc/build/apk_tests.gyp makes the build compile successfully from a Chromium+WebRTC configured checkout for Android APK tests.
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5919 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 12:59:49 +00:00
aluebs@webrtc.org
f927fd6481
Re-enable AGC tests:
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* AgcConfigTest.HasCorrectDefaultConfiguration
* AgcConfigTest.DealsWithInvalidParameters
* AgcConfigTest.CanGetAndSetAgcStatus
* AgcConfigTest.HasCorrectDefaultRxConfiguration
* AgcConfigTest.DealsWithInvalidRxParameters
* AgcConfigTest.CanGetAndSetRxAgcStatus
* AudioProcessingTest.AgcIsOnByDefault
* AudioProcessingTest.CanEnableAgcWithAllModes
* AudioProcessingTest.RxAgcShouldBeOffByDefault
* AudioProcessingTest.CanTurnOnDigitalRxAcg
* AudioProcessingTest.CannotTurnOnAdaptiveAnalogRxAgc
BUG=webrtc:2784
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12019006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5918 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 11:58:18 +00:00
kjellander@webrtc.org
7de47bce12
Remove use of tmpnam.
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This solves compilation with the Mac SDK 10.9.
BUG=3120, 3151
TEST=git try -t modules_tests:VideoProcessorIntegrationTest*
R=fischman@webrtc.org , henrike@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10739005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5917 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 08:04:26 +00:00
andrew@webrtc.org
2c3f1abb69
Replace flooding logs in rtp_sender.cc with a comment.
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Started occurring after:
https://webrtc-codereview.appspot.com/11129004
BUG=3153
R=andresp@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5916 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 21:26:34 +00:00