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Author SHA1 Message Date
henrik.lundin@webrtc.org
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
bjornv@webrtc.org
06c1d6f3a1 VoEVolumeTest: Adds error return tests. 
BUG=367
TESTED=trybots, voe_auto_test
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19469006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6139 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 09:03:33 +00:00
kwiberg@webrtc.org
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
pbos@webrtc.org
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
pbos@webrtc.org
4e2806d85f Remove WEBRTC_TRACE uses in video_engine/ 
Complements fixes by mflodman@.

BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6136 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 08:02:22 +00:00
kjellander@webrtc.org
98c76a120d Make vie/voe_auto_test accept non-supported flags without error. 
With the switch recipes on the buildbots and the deprecation of
the custom script at
https://code.google.com/p/webrtc/source/browse/trunk/webrtc/test/buildbot_tests.py
these tests will start failing when Chromium's runtest.py is passing
--brave-new-test-launcher --test-launcher-bot-mode
to the test.
A similar change was made for most of WebRTC's tests (that depends on
the test_support_main target) in
https://webrtc-codereview.appspot.com/2222005

BUG=chromium:346198
TEST=Successfully launched the executables on Linux and Mac using:
out/Release/voe_auto_test --brave-new-test-launcher --test-launcher-bot-mode --automated --test-launcher-summary-output=/tmp/tmpwhx6Zz
out/Release/vie_auto_test --brave-new-test-launcher --test-launcher-bot-mode --automated --capture_test_ensure_resolution_alignment_in_capture_device=false --test-launcher-summary-output=/tmp/tmpwhx6Zz

R=henrika@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 06:01:40 +00:00
henrike@webrtc.org
f048872e91 Adds a modified copy of talk/base to webrtc/base. It is the first step in 
migrating talk/base to webrtc/base.

BUG=N/A
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17479005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 18:00:26 +00:00
bjornv@webrtc.org
8d63d0ee70 Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255 
Rewritten the test to only check for valid volume when we have actually received a value from the audio device. To check if we have actually received a volume value is out of the scope for this test.

BUG=webrtc:367
TESTED=trybots
R=tina.legrand@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16499006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6123 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 14:14:56 +00:00
andresp@webrtc.org
93ec9c557b Revert "FieldTrial implementation for webrtc." (rev 6089) 
New wiring plans require it to be landed first in chrome for a cleaner roll of webrtc.

BUG=crbug/367114
R=tommi@webrtc.org
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6122 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 14:09:40 +00:00
asapersson@webrtc.org
e41dbee8a6 Reduced kMaxSampleDiffMs (limit to 22fps). 
BUG=1577
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6121 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 13:45:13 +00:00
pbos@webrtc.org
023b101f4e Move gflags usage to video_loopback. 
gflags aren't used by the test environment and is an unnecessary
dependency. They're only used by the video_loopback target, so moving
them there.

R=mflodman@webrtc.org
BUG=3113

Review URL: https://webrtc-codereview.appspot.com/12379006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6120 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 11:26:40 +00:00
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
perkj@webrtc.org
e9a604accd Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..." 
This breaks Chromium FYI builds and prevent roll of webrtc/libjingle to Chrome.

http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win%20Builder/builds/457


> Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
> 
> BUG=N/A
> R=andrew@webrtc.org, wu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/12199004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6116 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 08:15:48 +00:00
henrik.lundin@webrtc.org
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
stefan@webrtc.org
46e636a3f5 Fix failing test introduced with r6111. 
Test was assuming that getting the receive estimate of a stream which hasn't received packets would return an error, new behavior is to return 0.

TBR=wu@webrtc.org
BUG=crbug/371714

Review URL: https://webrtc-codereview.appspot.com/21419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6114 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 23:17:29 +00:00
stefan@webrtc.org
72885d1c91 Fixes log spam introduced with r6041. 
We shouldn't return an error if we don't yet have a valid estimate.

BUG=crbug/371714
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15469006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6111 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 22:09:27 +00:00
henrike@webrtc.org
2c7d1b39b9 Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base. 
BUG=N/A
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 18:03:09 +00:00
henrika@webrtc.org
6b02eea6ac Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6103 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 12:24:10 +00:00
henrika@webrtc.org
1cec3957b8 Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 12:19:19 +00:00
kwiberg@webrtc.org
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
henrika@webrtc.org
66021e0fa2 Removes parts of the webrtc::VoERTP_RTCP sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org, solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13489005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6100 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 08:53:27 +00:00
turaj@webrtc.org
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
turaj@webrtc.org
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
henrika@webrtc.org
3b76627afe Removes parts of the webrtc::VoEHardware sub API (relanding) 
Relanding https://webrtc-codereview.appspot.com/18399004/

TBR=niklase

Review URL: https://webrtc-codereview.appspot.com/16489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6092 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 11:43:00 +00:00
henrika@webrtc.org
3106b706c0 Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..." 
> Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
> 
> BUG=3206
> R=andrew@webrtc.org, niklas.enbom@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/18399004

TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6091 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 11:10:50 +00:00
henrika@webrtc.org
9de3d844ae Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=andrew@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6090 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 10:55:11 +00:00
andresp@webrtc.org
6a8a6723d3 FieldTrial implementation for webrtc. 
BUG=crbug/367114
R=asvitkine@chromium.org, mflodman@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 07:14:34 +00:00
wu@webrtc.org
02b286bfc9 Raise kViEMaxNumberOfChannels from 32 to 64 
Recent testing has shown that on modern desktops and laptops, decoding more than
32 low-resolution realtime video streams simultaneously is both possible and
desirable.

Reviewed:
https://webrtc-codereview.appspot.com/16449004/

TBR=mflodman
BUG=

Review URL: https://webrtc-codereview.appspot.com/17429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6087 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 22:22:41 +00:00
elham@webrtc.org
e37951d28f Updated WebRTC version to 3.53 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13489006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6081 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 17:09:31 +00:00
kwiberg@webrtc.org
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
wu@webrtc.org
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
braveyao@webrtc.org
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
fischman@webrtc.org
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
wu@webrtc.org
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
andrew@webrtc.org
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
andrew@webrtc.org
a0edf4cb04 Remove ALLOW_UNUSED. 
Turns out Chromium won't be applying this to COMPILE_ASSERT. We don't
need it at all then.

R=thakis@chromium.org
TBR=thakis@chromium.org

Review URL: https://webrtc-codereview.appspot.com/13469005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6055 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 18:18:02 +00:00
wu@webrtc.org
0224c20fa6 * Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter. 
* Re-enable test CaptureNtpTimeWithNetworkJitter.
* Use 100ms as the threadhold as a FYI since this is a performance test.

BUG=3271
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 17:42:43 +00:00
jiayl@webrtc.org
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
tina.legrand@webrtc.org
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
braveyao@webrtc.org
633aff6bd0 WebRTCDemo: correct set trace filter operation. 
BUG=3285
TEST=Manul Test
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6051 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 04:24:47 +00:00
andrew@webrtc.org
9f453b1a1b Add ALLOW_UNUSED and update COMPILE_ASSERT to Chromium's latest. 
Fixes building with gcc 4.8.

TBR=fdegans@google.com
BUG=chromium:321833

Review URL: https://webrtc-codereview.appspot.com/12439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-04 03:04:26 +00:00
jiayl@webrtc.org
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
henrika@webrtc.org
7f3a041d23 Removed NetworkTest.CanSwitchToExternalTransport since it tests an unsupported case and we should not maintain such a test. 
BUG=3289
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6043 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 13:59:58 +00:00
asapersson@webrtc.org
9205c87820 Pointers were not dereferenced in GetRtpStatistics. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9039005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6042 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 13:24:42 +00:00
stefan@webrtc.org
24bd364d3e Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels. 
This fixes an issue where the user doesn't know which channels are "active" and therefore can't properly sum the estimates for all channels.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6041 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 12:35:37 +00:00
andrew@webrtc.org
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
andrew@webrtc.org
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
turaj@webrtc.org
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
henrike@webrtc.org
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