48438c2c90
Enabling NetEq bit-exactness test for Win x64
...
A new reference file (neteq4_universal_ref_win_64.pcm) was generated and
uploaded.
Also removing the old hack to have different reference files
for different version of Visual Studio. The test is now only supporting
VS 2012 and later (_MSC_VER >= 1700). This makes the windows 32-bit
output identical to the generic reference file
(neteq4_universal_ref.pcm), so the specialized one
(neteq4_universal_ref_win_32.pcm) could have been removed. However,
since the resources sync mechanism does not include removing of old
files, a client could pick up the old reference and fail. Therefore,
this cl also updates neteq4_universal_ref_win_32.pcm to be identical to
neteq4_universal_ref.pcm.
BUG=1458
R=kjellander@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14569005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6204 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 16:07:43 +00:00
125ffd709d
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
...
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 15:20:44 +00:00
70bb2d5755
Revert r6198 "Expose the original packet length in in the RTP play tools."
...
TBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6200 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 13:25:48 +00:00
e208458643
Expose the original packet length in in the RTP play tools.
...
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6198 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 13:09:16 +00:00
be4ab99a53
Disabling RealFFTTest.RealAndComplexMatch and AudioProcessingTest.Formats as they currently are broken with gcc 4.8.
...
BUG=3370
R=bjornv@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6197 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 12:42:01 +00:00
a36db970bd
Suppress GMOCK printouts from TestVideoSenderWithVp8
...
Adding a missing EXPECT_CALL.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20529005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6196 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 11:16:10 +00:00
a826006132
Add NACK and RPSI packet types to RTCP packet builder.
...
Fixes bug found when parsing received RPSI packet.
BUG=2450
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6194 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 09:53:51 +00:00
cb711f77d2
Add interface to propagate audio capture timestamp to the renderer.
...
BUG=3111
R=andrew@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6189 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 17:39:11 +00:00
ebb467fdc8
Avoid NACK-list flush error on keyframe packets.
...
Receiver code used to indicate a flush error even if the incoming packet
is a keyframe, forcing a request of a keyframe. Now it takes this
keyframe into account and doesn't error as the stream is decodable from
this point.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15549005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6188 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 15:28:02 +00:00
64339a7069
Don't crash if a frame returned from the decoder is too old.
...
BUG=crbug/371805
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6187 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 13:31:35 +00:00
725e582461
Use the new gyp_var_prefix local variable set by gyp instead of the
...
global GYP_VAR_PREFIX set by the makefiles, since the latter is not
guaranteed to still be the same value at the time the command is
executed. Also, use abspath instead of realpath to convert paths to
absolute, since realpath expands to the empty string if the target file
doesn't exist, complicating build debugging.
BUG=
R=andrew@webrtc.org , torne@chromium.org
Review URL: https://webrtc-codereview.appspot.com/12559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6186 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 17:56:10 +00:00
14abcc7322
libvpx's UNUSED macro conflicts with webrtc/base's. Added missing include of assert.h. Globally defined function "Unused" in talk/base and its copy (webrtc/base) is causing a conflict.
...
libvpx macro (UNUSED) can be found here:
http://src.chromium.org/viewvc/chrome/trunk/deps/third_party/libvpx/source/libvpx/vpx/vpx_codec.h
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6185 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 16:54:44 +00:00
a3b5673879
common_audio/signal_processing: Removes macro WEBRTC_SPL_UMUL_RSFT16
...
This macro was only used on two lines in iSACfix and I replaced those with the operations the macro performed.
BUG=3348
TESTED=trybots, manual unittests
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6184 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 12:11:20 +00:00
1b21a57902
common_audio/signal_processing: Removed macro WEBRTC_SPL_SUB_SAT_W16
...
Macro was only mapping a function used in one place.
BUG=3348
TESTED=trybots, unittests
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6180 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 06:40:31 +00:00
d5da25063c
Revert "Revert "Audio processing: Feed each processing step its choice
...
of int or float data"
This reverts commit 6142.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6172 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-15 11:17:21 +00:00
b4e80e095f
Re-enable almost all NetEqDecodingTests for Android
...
All but three tests in NetEqDecodingTest could be re-enabled without
any changes. Also making sure that the TestNetworkStatistics test exits
on first diff. (Otherwise, the log output gets flooded with error
messages.)
The tests that are still disabled are:
NetEqDecodingTest.TestBitExactness
NetEqDecodingTest.TestNetworkStatistics
NetEqDecodingTest.DecoderError
BUG=3343
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6168 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-15 07:14:00 +00:00
7cb4752184
WebRTCDemo: couldn't run a second time. The reason is voe could register/unregister for each run, but vie would expect initialization only once per process.
...
This cl is to teach videocapture android how to deinitialize and allow it to be re-initializable.
BUG=3284
TEST=ManualTest
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6167 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-15 03:18:15 +00:00
21299d4e00
Remove the use of AudioFrame::energy_ from AudioProcessing and VoE.
...
We want to remove energy_ entirely as we've seen that carrying around
this potentially invalid value is dangerous.
Results in the removal of AudioBuffer::is_muted(). This wasn't used in
practice any longer, after the level calculation moved directly to
channel.cc
Instead, now use ProcessMuted() in channel.cc, to shortcut the level
computation when the signal is muted.
BUG=3315
TESTED=Muting the channel in voe_cmd_test results in rms=127.
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6159 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 19:00:59 +00:00
88abf11cad
Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe.
...
BUG=3111
TEST=try bots
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6152 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 16:53:51 +00:00
a36ad6929d
Add webrtc field trials API.
...
From now on it is expected that code linking system_wrappers.gyp:system_wrappers
provides an implementation for field_trial API or links with the default one in
system_wrappers.gyp:field_trial_default.
Note: Since there is no use of webrtc::field_trial API inside webrtc this CL on
itself does not forces the clients to actually define it. It however lays the
API and updates the gyp rules to link with so that it is ready to use.
Tested: Introduced a use of field trial in system wrappers and make sure all
bots were building successfully.
BUG=crbug/367114
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6147 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 12:24:04 +00:00
2fa17015d1
Re-enable NetEqExternalDecoderTest for Android
...
The test runs without problems now.
BUG=3343
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16519005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6144 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 11:45:22 +00:00
bf93fb3176
Re-enable NetEQ DecoderDatabase test for Android
...
The test was failing because iLBC is not enabled on Android. Now, the
test is using PCM16B instead.
BUG=3343
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6143 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 10:42:03 +00:00
b1a66d166c
Revert "Audio processing: Feed each processing step its choice of int or float data"
...
This reverts r6138.
tbr=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6142 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 09:39:56 +00:00
db60434b31
Re-enable the BitrateEstimatorTest cases for the Call API.
...
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6141 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 09:15:19 +00:00
5c49c64de5
Remove all use of AudioFrame::energy_ from AudioCodingModule
...
Since r6117, the energy is always calculated in the mixer module,
regardless of the value that ACM sets for energy_.
This part of the the aftermath of issue 3255.
BUG=3255
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6140 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 09:06:52 +00:00
934a265a47
Audio processing: Feed each processing step its choice of int or float data
...
Each audio processing step is given a pointer to an AudioBuffer, where
it can read and write int data. This patch adds corresponding
AudioBuffer methods to read and write float data; the buffer will
automatically convert the stored data between int and float as
necessary.
This patch also modifies the echo cancellation step to make use of the
new methods (it was already using floats internally; now it doesn't
have to convert from and to ints anymore).
(The reference data to the ApmTest.Process test had to be modified
slightly; this is because the echo canceller no longer unnecessarily
converts float data to int and then immediately back to float for each
iteration in the loop in EchoCancellationImpl::ProcessCaptureAudio.)
BUG=
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18399005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6138 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 09:01:35 +00:00
3d5cb33da4
Remove WEBRTC_TRACE use in video_capture/
...
Does not touch platform-specific code.
BUG=3153
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6137 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 08:42:07 +00:00
c3e8abda7c
Deleting all NetEq3 files
...
NetEq3 is deprecated and replaced by NetEq4
(webrtc/modules/audio_coding/neteq4/).
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14469007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6118 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 10:40:52 +00:00
4d363ae305
The webrtc::AudioFrame struct contains a variable energy_. Since the energy isn't always calculated when the frame is created, this change makes the CalculateEnergy method in Audio Conference Mixer always calculate the energy.
...
This part of the the aftermath of issue 3255.
BUG=3255
R=andrew@webrtc.org , henrike@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6117 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 08:50:02 +00:00
3a5825909d
Deleting all ACM1 files
...
ACM1 is deprecated and replaced by ACM2
(webrtc/modules/audio_coding/acm2/).
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18429005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6115 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 08:08:56 +00:00
924e81f797
Echo cancellation functions docs: Follow style guide w.r.t. placement of *
...
The style guide says to use "void* x", not void *x", and the code in
these files already do so, but the comments do not. Fix that.
Also, in the interest of reducing eye strain, I fixed the vertical
alignment in a small number of cases.
BUG=
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6101 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-12 09:55:19 +00:00
b9863ce6ba
One of the NetEq methods needs to be virtual.
...
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6099 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-10 00:58:49 +00:00
17bf9a2c5e
Modifying neteq.gyp
...
|codecs| list is introduced, which is used in both |neteq_dependencies| and AudiDecoder unittests dependencies. This way, AudioDecoder unittests depend only on |codecs| and not on the entire |neteq_dependencies|, which is unnecessary.
TEST=trybots
BUG=
R=andrew@webrtc.org , henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6094 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-09 18:04:50 +00:00
4cc763621e
AudioBuffer: Eliminate data_was_mixed_, and document what's left of data_
...
data_was_mixed_ was always false, so it can be removed. That makes the
role of data_ simpler, but not so simple that it doesn't merit an
explanation.
BUG=
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6076 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-08 07:10:11 +00:00
66773a032a
Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine.
...
BUG=3111
TEST=try bots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6074 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-07 17:09:44 +00:00
94f1d4cd55
Fix odd codes in video_capture on Mac.
...
BUG=3272
TEST=vie_auto_test
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6070 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-07 02:57:13 +00:00
b1eb43142e
video_render.gypi: clean up some libraries directives to be more specific.
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6068 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-07 00:09:30 +00:00
ed4cb56575
Remove timestamp_extrapolator's dependency to Clock and vcm defines.
...
TEST=existing tests
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6058 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-06 04:50:49 +00:00
382c0c209d
Allow the RTP level indicator computation to work at any sample rate.
...
Break out the computation to a separate class, and call directly into
this from channel.cc rather than going through AudioProcessing. This
circumvents AudioProcessing's sample rate limitations.
We now compute the RMS over all samples rather than downmixing to a
single channel. This makes the call point in channel.cc easier, is
more "correct" and should have similar (negligible) complexity.
This caused slight changes in the RMS output, so the ApmTest.Process
reference has been updated. Snippet of the failing output:
[ RUN ] ApmTest.Process
Running test 4 of 12...
Value of: rms_level
Actual: 27
Expected: test->rms_level()
Which is: 28
Running test 5 of 12...
Value of: rms_level
Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 6 of 12...
Value of: rms_level
Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 10 of 12...
Value of: rms_level
Actual: 27
Expected: test->rms_level()
Which is: 28
Running test 11 of 12...
Value of: rms_level
Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 12 of 12...
Value of: rms_level
Actual: 26
Expected: test->rms_level()
Which is: 27
BUG=3290
TESTED=Chrome assert is avoided and both voe_cmd_test and apprtc
produce reasonable printed out results from RMS().
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6056 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-05 18:22:21 +00:00
4220434d37
Implement the Windows screen capturer using the Magnification API.
...
The original ScreenCapturerWin is renamed ScreenCapturerWinGdi.
BUG=2789
TESTED=full desktop cast and single monitor cast works on win7 and win8 desktop mode. Have to use GDI capturer on win8 metro mode. Changing display configuration work on the fly.
R=sergeyu@chromium.org , wez@chromium.org
Committed: https://code.google.com/p/webrtc/source/detail?r=6048
Review URL: https://webrtc-codereview.appspot.com/12149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6053 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-05 16:08:47 +00:00
7dccce3948
Revert 6048 "Implement the Windows screen capturer using the Mag..."
...
> Implement the Windows screen capturer using the Magnification API.
> The original ScreenCapturerWin is renamed ScreenCapturerWinGdi.
>
> BUG=2789
> TESTED=full desktop cast and single monitor cast works on win7 and win8 desktop mode. Have to use GDI capturer on win8 metro mode. Changing display configuration work on the fly.
> R=sergeyu@chromium.org , wez@chromium.org
>
> Review URL: https://webrtc-codereview.appspot.com/12149004
TBR=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15429005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6052 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-05 11:17:26 +00:00
b235c56017
Implement the Windows screen capturer using the Magnification API.
...
The original ScreenCapturerWin is renamed ScreenCapturerWinGdi.
BUG=2789
TESTED=full desktop cast and single monitor cast works on win7 and win8 desktop mode. Have to use GDI capturer on win8 metro mode. Changing display configuration work on the fly.
R=sergeyu@chromium.org , wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/12149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6048 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-03 00:16:29 +00:00
e44a84d851
Only clamp to 16 kHz when AECM is enabled.
...
Otherwise we could needlessly downsample to 16 kHz (rather than 32 kHz)
when HW AEC is used.
BUG=3259
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6033 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-30 18:58:23 +00:00
65f933899b
Fix constness of AudioBuffer accessors.
...
Don't return non-const pointers from const accessors and deal with the
spillover. Provide overloaded versions as needed.
Inspired by kwiberg:
https://webrtc-codereview.appspot.com/12379005/
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6030 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-30 16:44:13 +00:00
9bd49becc1
Fix a data race in ACM1 when audio is pulled.
...
BUG=chromium:348511
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6026 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 20:27:45 +00:00
f2aafe4355
Added include of assert.h for files calling assert but missing the include.
...
BUG=N/A
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19409005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6022 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 17:54:17 +00:00
ceffdbc371
Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord()
...
R=henrike@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6018 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 17:11:07 +00:00
0300939484
Disable failing GoogleWifiTrace3Mbps.
...
Disables BweFeedbackTest.GoogleWifiTrace3Mbps instead of
BweSimulation.GoogleWifiTrace3Mbps.
TBR=stefan@webrtc.org
BUG=3277
Review URL: https://webrtc-codereview.appspot.com/20389005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6017 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 15:25:59 +00:00
9353e6bc55
Disable GoogleWifiTrace3Mbps.
...
Breaks bots, according to stefan@ there's a missing file for this test
to run.
TBR=stefan@webrtc.org
BUG=3277
Review URL: https://webrtc-codereview.appspot.com/13409005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6016 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 14:49:56 +00:00
dfe2a1c995
Adding BweFeedbackTest which tracks BWE performance over a set of simulated scenarios.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11019006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6015 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 14:21:42 +00:00
