9085 Commits

Author SHA1 Message Date
ossu
a97c5d233d Add ossu@ to OWNERS of audio/ and modules/audio_coding/
BUG=none

Review-Url: https://codereview.webrtc.org/2576213003
Cr-Commit-Position: refs/heads/master@{#15640}
2016-12-15 15:52:14 +00:00
mflodman
d79f97b542 Fixing loopback video test by reconfiguring the encoder to correct size.
Same as https://codereview.webrtc.org/2480753002, but with a small fix.

BUG=none

Review-Url: https://codereview.webrtc.org/2578143002
Cr-Commit-Position: refs/heads/master@{#15639}
2016-12-15 15:24:38 +00:00
philipel
721d402d71 Create VideoReceiver with external VCMTiming object.
In order for the VCMTiming object to be correctly updated with decoding timings
when running the WebRTC-NewVideoJitterBuffer experiment the VCMTiming object
has to be available in both the VideoReceiver and the video_coding::FrameBuffer
class. Therefore the VCMTiming object is created in VideoRecieveStream and
then passed to VideoReceiver/video_coding::FrameBuffer as they are constructed.

BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2575473004
Cr-Commit-Position: refs/heads/master@{#15638}
2016-12-15 15:11:01 +00:00
henrika
ac8d5164f0 Improves release of allocated audio resources on Android.
BUG=webrtc:6890
R=magjed@webrtc.org, solenberg@webrtc.org

Review-Url: https://codereview.webrtc.org/2574053003 .
Cr-Commit-Position: refs/heads/master@{#15637}
2016-12-15 14:43:10 +00:00
terelius
43c382111d Revert of Avoid precision loss in TrendlineEstimator from int64_t -> double conversion (patchset #7 id:120001 of https://codereview.webrtc.org/2577463002/ )
Reason for revert:
Multiple definitions of TestEstimator

Original issue's description:
> Avoid precision loss in TrendlineEstimator by passing the arrival time as an int64_t instead of a double.
>
> BUG=webrtc:6884
>
> Committed: https://crrev.com/c12cbaf9dd0729dd45f3fc45a1938d1b3455e40a
> Cr-Commit-Position: refs/heads/master@{#15631}

TBR=stefan@webrtc.org,brandtr@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6884

Review-Url: https://codereview.webrtc.org/2582513002
Cr-Commit-Position: refs/heads/master@{#15636}
2016-12-15 14:42:50 +00:00
terelius
0bac07a89b Revert of Avoid precision loss in MedianSlopeEstimator from int64_t -> double conversion (patchset #3 id:40001 of https://codereview.webrtc.org/2578543002/ )
Reason for revert:
Multiple definitions of TestEstimator

Original issue's description:
> Pass arrival time as an int64_t rather than a double to the MedianSlopeEstimator to avoid precision loss.
>
> Also clean up the unit test.
>
> BUG=webrtc:6892
>
> Committed: https://crrev.com/ebcbcc3b2451f5c4fb07f7b37815bd54f364d057
> Cr-Commit-Position: refs/heads/master@{#15634}

TBR=brandtr@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6892

Review-Url: https://codereview.webrtc.org/2572353003
Cr-Commit-Position: refs/heads/master@{#15635}
2016-12-15 14:41:43 +00:00
terelius
ebcbcc3b24 Pass arrival time as an int64_t rather than a double to the MedianSlopeEstimator to avoid precision loss.
Also clean up the unit test.

BUG=webrtc:6892

Review-Url: https://codereview.webrtc.org/2578543002
Cr-Commit-Position: refs/heads/master@{#15634}
2016-12-15 14:31:23 +00:00
nisse
df2ceb88a8 Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ )
Reason for revert:
Fixing perf tests.

Original issue's description:
> Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ )
>
> Reason for revert:
> Crashes perf tests, e.g.,
>
> ./out/Debug/webrtc_perf_tests --gtest_filter='FullStackTest.ScreenshareSlidesVP8_2TL_VeryLossyNet'
>
> dies with an assert related to rtc::Optional.
>
> Original issue's description:
> > Delete VideoFrame default constructor, and IsZeroSize method.
> >
> > This ensures that the video_frame_buffer method never can return a
> > null pointer.
> >
> > BUG=webrtc:6591
> >
> > Committed: https://crrev.com/bfcf561923a42005e4c7d66d8e72e5932155f997
> > Cr-Commit-Position: refs/heads/master@{#15574}
>
> TBR=magjed@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6591
>
> Committed: https://crrev.com/0989fbcad2ca4eb5805a77e8ebfefd3af06ade23
> Cr-Commit-Position: refs/heads/master@{#15597}

TBR=magjed@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6591

Review-Url: https://codereview.webrtc.org/2574183002
Cr-Commit-Position: refs/heads/master@{#15633}
2016-12-15 14:30:00 +00:00
michaelt
bf65be5435 Wire-up audio BWE with overhead.
BUG=webrtc:6638

Review-Url: https://codereview.webrtc.org/2565353002
Cr-Commit-Position: refs/heads/master@{#15632}
2016-12-15 14:24:52 +00:00
terelius
c12cbaf9dd Avoid precision loss in TrendlineEstimator by passing the arrival time as an int64_t instead of a double.
BUG=webrtc:6884

Review-Url: https://codereview.webrtc.org/2577463002
Cr-Commit-Position: refs/heads/master@{#15631}
2016-12-15 14:20:03 +00:00
hbos
3168c7a04b Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time.
The spec renamed these recently:
https://w3c.github.io/webrtc-stats/#since-21-sep-2016*

BUG=chromium:627816

Review-Url: https://codereview.webrtc.org/2576383002
Cr-Commit-Position: refs/heads/master@{#15630}
2016-12-15 14:17:15 +00:00
nisse
6a58f33450 Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2581663002/ )
Reason for revert:
This change broke Chrome too. It's stats processing wants to make a copy of webrtc's stats mapping, which is no longer possible with std::unique_ptr.

Original issue's description:
> Reland of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2576673002/ )
>
> Reason for revert:
> Downstream project fixed to not make copies while iterating over the stats mapping.
>
> Original issue's description:
> > Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2567143003/ )
> >
> > Reason for revert:
> > The change from rtc::linked_ptr to std::unique_ptr broke a downstream project.
> >
> > Original issue's description:
> > > Delete rtc::linked_ptr. Only use, in statstypes.h, replaced with std::unique_ptr.
> > >
> > > BUG=webrtc:6424
> > >
> > > Committed: https://crrev.com/36f74e55792cae19db8b222c29aa38d6e0eb1225
> > > Cr-Commit-Position: refs/heads/master@{#15588}
> >
> > TBR=solenberg@webrtc.org,pthatcher@webrtc.org,hta@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:6424
> >
> > Committed: https://crrev.com/8afbc8cba65d99bb7a0feece8fb3055b144106b1
> > Cr-Commit-Position: refs/heads/master@{#15589}
>
> TBR=solenberg@webrtc.org,pthatcher@webrtc.org,hta@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:6424
>
> Committed: https://crrev.com/06035cf53abad80b0525f286a3b81e388cc7ee00
> Cr-Commit-Position: refs/heads/master@{#15627}

TBR=solenberg@webrtc.org,pthatcher@webrtc.org,hta@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2579753002
Cr-Commit-Position: refs/heads/master@{#15629}
2016-12-15 11:54:52 +00:00
hbos
7bf5369763 RTCStatsIntegrationTest: TestMemberIsIDReference on all defined IDs.
This makes sure that the referenced stats dictionaries exist.

BUG=chromium:627816

Review-Url: https://codereview.webrtc.org/2577033002
Cr-Commit-Position: refs/heads/master@{#15628}
2016-12-15 11:33:42 +00:00
nisse
06035cf53a Reland of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2576673002/ )
Reason for revert:
Downstream project fixed to not make copies while iterating over the stats mapping.

Original issue's description:
> Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2567143003/ )
>
> Reason for revert:
> The change from rtc::linked_ptr to std::unique_ptr broke a downstream project.
>
> Original issue's description:
> > Delete rtc::linked_ptr. Only use, in statstypes.h, replaced with std::unique_ptr.
> >
> > BUG=webrtc:6424
> >
> > Committed: https://crrev.com/36f74e55792cae19db8b222c29aa38d6e0eb1225
> > Cr-Commit-Position: refs/heads/master@{#15588}
>
> TBR=solenberg@webrtc.org,pthatcher@webrtc.org,hta@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6424
>
> Committed: https://crrev.com/8afbc8cba65d99bb7a0feece8fb3055b144106b1
> Cr-Commit-Position: refs/heads/master@{#15589}

TBR=solenberg@webrtc.org,pthatcher@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2581663002
Cr-Commit-Position: refs/heads/master@{#15627}
2016-12-15 11:12:22 +00:00
Henrik Kjellander
0b571a11a4 MB: Enable memcheck for the linux_memcheck trybot.
This was missed in https://codereview.webrtc.org/2510033004

BUG=chromium:497757
NOTRY=True
TBR=ehmaldonado@webrtc.org

Review-Url: https://codereview.webrtc.org/2577903003 .
Cr-Commit-Position: refs/heads/master@{#15626}
2016-12-15 10:09:17 +00:00
hbos
e10e6d1f47 RTCOutboundRTPStreamStats.roundTripTime: Only report non-negative values.
Underlying stats gatherers may otherwise default it to -1.

BUG=chromium:669877, chromium:627816

Review-Url: https://codereview.webrtc.org/2562703007
Cr-Commit-Position: refs/heads/master@{#15625}
2016-12-15 09:54:38 +00:00
hta
41286496cb Move all codec specific definitions from modules_include
This CL moves all codec specific definitions into their own
header files in the respective codec directory, and replaces
it with an include in the top level directory.

This is to facilitate getting the code out of the header files.

No behavioral changes are expected.

BUG=webrtc:6842

Review-Url: https://codereview.webrtc.org/2555993003
Cr-Commit-Position: refs/heads/master@{#15623}
2016-12-15 08:54:15 +00:00
zijiehe
2769ec62c0 Add WriteIsolatedOutput() functions
WriteIsolatedOutput() functions write large content into swarming isolated
output folder, which are useful to log large test data for debugging purpose.

BUG=webrtc:6732

TBR=holmer@webrtc.org

Review-Url: https://codereview.webrtc.org/2558693002
Cr-Commit-Position: refs/heads/master@{#15616}
2016-12-14 23:03:11 +00:00
hta
88cf05cf73 Guard against uninitialized packetization modes.
This change inserts a RTC_CHECK for illegal packetization modes
when RTP packetizers are constructed.

This should help find places where this field is not initialized.

BUG=webrtc:6858

Review-Url: https://codereview.webrtc.org/2575073002
Cr-Commit-Position: refs/heads/master@{#15614}
2016-12-14 20:48:39 +00:00
hbos
9a394f0649 Skip RTCMediaStreamTrackStats.echoReturnLoss[Enhancement] default value.
Due to the Chromium implementation[1] of GetAudioProcesssingStats,
echoReturnLoss and echoReturnLossEnhancement could default to -100 when
no value was available. This should be improved by using rtc::Optional
or AudioProcessorInterface::GetStats being able to return false, but
this requires a bunch of refactoring.

In the meantime we "blacklist" the value -100 which is a nonsense value
anyway. In that case echoReturnLoss[Enhancement] is correctly left
undefined.

[1] https://cs.chromium.org/chromium/src/content/renderer/media/media_stream_audio_processor_options.cc?sq=package:chromium&dr=C&rcl=1481530670&l=461

BUG=chromium:669877

Review-Url: https://codereview.webrtc.org/2573443002
Cr-Commit-Position: refs/heads/master@{#15611}
2016-12-14 15:58:30 +00:00
henrika
c3c2f31852 Adds basic Bluetooth support to AppRTCMobile
BUG=webrtc:6649

- Supports Bluetooth Headset profile.
- Detects new BT headset:
  + enabled at start, and
  + powered on during active call.
- Enables/disables BT SCO channel when BT device is selected.
- Removes proximity sensor usage to avoid conflicts (will be added again later).
- Adds new (unused) APIs to explicitly select audio device.
- Starts routing audio to BT headset when enabled, i.e, BT is default.

Review-Url: https://codereview.webrtc.org/2501983002
Cr-Commit-Position: refs/heads/master@{#15610}
2016-12-14 15:37:01 +00:00
johan
db8af2a953 Run 'git cl format --full' on Base64.
# Legal requires us to keep the original license header.
NOPRESUBMIT=true
BUG=None

Review-Url: https://codereview.webrtc.org/2574143002
Cr-Commit-Position: refs/heads/master@{#15609}
2016-12-14 15:15:19 +00:00
danilchap
9006987243 Remove deprecated RTPSender::SendPadData
BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2551143004
Cr-Commit-Position: refs/heads/master@{#15608}
2016-12-14 14:16:43 +00:00
johan
e2ec7c270b Remove static cast from H264SpropParameterSets.
This CL is chained to https://codereview.webrtc.org/2539153002/ .

BUG=webrtc:5948

Review-Url: https://codereview.webrtc.org/2568953005
Cr-Commit-Position: refs/heads/master@{#15607}
2016-12-14 14:08:38 +00:00
ivoc
930959d261 Improvements to the reliability of the echo detector perf test.
BUG=chromium:673683

Review-Url: https://codereview.webrtc.org/2568883004
Cr-Commit-Position: refs/heads/master@{#15606}
2016-12-14 13:48:25 +00:00
johan
8fc0c4c32f Add vector<uint8_t> to Base64 decoded data types.
This is a prerequisite to decode fmtp sprop-parameter-sets into
the right encoding for H264SpsPpsTracker.

# Legal requires us to keep the original license header.
NOPRESUBMIT=true
BUG=webrtc:5948

Review-Url: https://codereview.webrtc.org/2539153002
Cr-Commit-Position: refs/heads/master@{#15604}
2016-12-14 12:13:05 +00:00
aleloi
a5bb562bfe Delete webrtc/transport.h.
Preliminary work for this deletion was done in
https://codereview.webrtc.org/2426563003. The header has been moved to
webrtc/api/transport.h. This follow-up CL may cause breakages for
clients that have not updated their includes.

TBR=kwiberg@webrtc.org
BUG=webrtc:6785

Review-Url: https://codereview.webrtc.org/2568953007
Cr-Commit-Position: refs/heads/master@{#15601}
2016-12-14 11:48:53 +00:00
peah
9e1e6c599d Corrected access of null pointer in audioproc_f:
The previous CL that added the ability to add
and artificial nearend signal had an issue with
null pointer access.

This is addressed in this CL.

BUG=webrtc:6018

Review-Url: https://codereview.webrtc.org/2573033003
Cr-Commit-Position: refs/heads/master@{#15600}
2016-12-14 11:12:27 +00:00
henrika
63e6a38745 Removes verification of audio parameters on Android
TBR=glaznev
BUG=webrtc:6890

Review-Url: https://codereview.webrtc.org/2572963003
Cr-Commit-Position: refs/heads/master@{#15599}
2016-12-14 10:53:31 +00:00
nisse
0989fbcad2 Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ )
Reason for revert:
Crashes perf tests, e.g.,

./out/Debug/webrtc_perf_tests --gtest_filter='FullStackTest.ScreenshareSlidesVP8_2TL_VeryLossyNet'

dies with an assert related to rtc::Optional.

Original issue's description:
> Delete VideoFrame default constructor, and IsZeroSize method.
>
> This ensures that the video_frame_buffer method never can return a
> null pointer.
>
> BUG=webrtc:6591
>
> Committed: https://crrev.com/bfcf561923a42005e4c7d66d8e72e5932155f997
> Cr-Commit-Position: refs/heads/master@{#15574}

TBR=magjed@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6591

Review-Url: https://codereview.webrtc.org/2574123002
Cr-Commit-Position: refs/heads/master@{#15597}
2016-12-14 10:06:49 +00:00
ivoc
7b2516620c Fix for left shift of negative value in NetEq.
BUG=chromium:666612

Review-Url: https://codereview.webrtc.org/2569193002
Cr-Commit-Position: refs/heads/master@{#15596}
2016-12-14 10:00:02 +00:00
nisse
bd6c6fa309 Delete method Pathname::url and base/urlencode*
BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2572703002
Cr-Commit-Position: refs/heads/master@{#15595}
2016-12-14 09:43:08 +00:00
skvlad
bb66ec3573 Disable flaky test VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9
This test is flaky on all platforms, not just Android. Disabling it entirely until webrtc:6057 is fixed.

BUG=webrtc:6057

Review-Url: https://codereview.webrtc.org/2568743007
Cr-Commit-Position: refs/heads/master@{#15594}
2016-12-14 09:17:34 +00:00
peah
e0eae3cec6 This CL adds the basic framework for AEC3 in the audio processing module.
It will be followed by a number of other CLs that extends this framework.

BUG=webrtc:6018

Review-Url: https://codereview.webrtc.org/2567513003
Cr-Commit-Position: refs/heads/master@{#15593}
2016-12-14 09:16:28 +00:00
nisse
db397429d4 Delete unused class rtc::RegKey.
BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2571763002
Cr-Commit-Position: refs/heads/master@{#15592}
2016-12-14 08:30:33 +00:00
nisse
43c5a974b4 Delete stl_util.h. Unused since cl https://codereview.webrtc.org/2447103002
BUG=None

Review-Url: https://codereview.webrtc.org/2570833002
Cr-Commit-Position: refs/heads/master@{#15590}
2016-12-14 08:07:44 +00:00
nisse
8afbc8cba6 Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2567143003/ )
Reason for revert:
The change from rtc::linked_ptr to std::unique_ptr broke a downstream project.

Original issue's description:
> Delete rtc::linked_ptr. Only use, in statstypes.h, replaced with std::unique_ptr.
>
> BUG=webrtc:6424
>
> Committed: https://crrev.com/36f74e55792cae19db8b222c29aa38d6e0eb1225
> Cr-Commit-Position: refs/heads/master@{#15588}

TBR=solenberg@webrtc.org,pthatcher@webrtc.org,hta@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2576673002
Cr-Commit-Position: refs/heads/master@{#15589}
2016-12-14 08:06:38 +00:00
nisse
36f74e5579 Delete rtc::linked_ptr. Only use, in statstypes.h, replaced with std::unique_ptr.
BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2567143003
Cr-Commit-Position: refs/heads/master@{#15588}
2016-12-14 07:36:26 +00:00
Henrik Kjellander
a5073c0c97 Disable AudioDeviceTest.StartPlayoutOnTwoInstances on iOS
This test fails when running on real iOS devices.

BUG=webrtc:6889
TBR=henrika@webrtc.org

Review-Url: https://codereview.webrtc.org/2576653002 .
Cr-Commit-Position: refs/heads/master@{#15586}
2016-12-14 06:25:24 +00:00
deadbeef
62802a1b0e Fixing possible crash due to RefCountedChannel assignment operator.
We relied on the default destructor of RefCountedChannel to destroy its
members in reverse initialization order (deleting the DTLS wrapper
before the underlying ICE channel).

However, std::vector also may use the default assignment operator, which
performs a member-wise copy in initialization order. Which results in
deleting the ICE channel before the DTLS one. This CL fixes this by
using a vector of pointers instead of structures, and uses RefCountedObject
to handle ref-counting.

BUG=chromium:672951

Review-Url: https://codereview.webrtc.org/2571683004
Cr-Commit-Position: refs/heads/master@{#15583}
2016-12-14 00:38:46 +00:00
deadbeef
b236257763 Fixing integer overflow when parsing bandwidth attribute.
It's still valid SDP so just clamp it at INT_MAX.

BUG=chromium:648071

Review-Url: https://codereview.webrtc.org/2571073002
Cr-Commit-Position: refs/heads/master@{#15582}
2016-12-14 00:37:16 +00:00
gyzhou
95aa96465d Support external audio mixer in webrtc 2.
An external audio mixer will be passed from PeerConnectionFactory to
AudioTransportProxy.

This CL has rewritten based on reverted CL
https://codereview.chromium.org/2539213003/
The only difference is that
  static MediaEngineInterface* Create(
      webrtc::AudioDeviceModule* adm,
      const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
          audio_decoder_factory,
      WebRtcVideoEncoderFactory* video_encoder_factory,
      WebRtcVideoDecoderFactory* video_decoder_factory);
in media/engine/webrtcmediaengine.h is kept in this CL instead of
replaced for backward compatibility.

BUG=webrtc:6457

Review-Url: https://codereview.webrtc.org/2570993002
Cr-Commit-Position: refs/heads/master@{#15580}
2016-12-13 22:06:35 +00:00
deadbeef
7af91ddd6b Removing "crypto_required" from MediaContentDescription.
"Crypto required" is a property of the PeerConnection of construction
time; it has nothing to do with SDP. So I'm moving it out of
MediaContentDescription and putting it in the BaseChannel constructor
instead. This is more intuitive, and provides the added assurance that
"secure_required_" can't be flipped from "true" to "false".

BUG=None

Review-Url: https://codereview.webrtc.org/2537343003
Cr-Commit-Position: refs/heads/master@{#15579}
2016-12-13 19:29:16 +00:00
hnsl
b68cc75f19 ParseCandidate(): Refactor to fix memcheck false positive.
Also make supported protocols explicit in check.

Fix inconsistency where TLS_PROTOCOL_NAME was not exported.

BUG=webrtc:6885

Review-Url: https://codereview.webrtc.org/2570803003
Cr-Commit-Position: refs/heads/master@{#15577}
2016-12-13 18:33:47 +00:00
minyue
301fc4a712 Update common_audio/smoothing_filter.
The improvement is mainly to extrapolate missing samples so that when querying the output, it assumes the input to continue even if no actual new samples are added.

The new implementation does not rely on base/exp_filter any longer. This is because it would be a bit cumbersome. base/exp_filter does pre-extrapolate, i.e., it assumes the all missing samples since the last sample equals the new sample.

BUG=webrtc:6443

Review-Url: https://codereview.webrtc.org/2551363002
Cr-Commit-Position: refs/heads/master@{#15575}
2016-12-13 14:53:07 +00:00
nisse
bfcf561923 Delete VideoFrame default constructor, and IsZeroSize method.
This ensures that the video_frame_buffer method never can return a
null pointer.

BUG=webrtc:6591

Review-Url: https://codereview.webrtc.org/2541863002
Cr-Commit-Position: refs/heads/master@{#15574}
2016-12-13 14:08:39 +00:00
kthelgason
46711db355 Disable flaky QualityScaler tests for now.
BUG=webrtc:6799
TBR=sprang@webrtc.org

Review-Url: https://codereview.webrtc.org/2564423002
Cr-Commit-Position: refs/heads/master@{#15573}
2016-12-13 13:32:31 +00:00
hnsl
277b250936 Refactor "secure bool" into explicit PROTO_TLS.
BUG=none

Review-Url: https://codereview.webrtc.org/2568833002
Cr-Commit-Position: refs/heads/master@{#15572}
2016-12-13 13:17:31 +00:00
thomasanderson
ef16e9960f Add a gtk3 port of peerconnection_client on Linux
BUG=668446

Review-Url: https://codereview.webrtc.org/2563203002
Cr-Commit-Position: refs/heads/master@{#15569}
2016-12-13 10:57:50 +00:00
palmkvist
349092befe Logging basic bad call detection
BUG=webrtc:6814

Review-Url: https://codereview.webrtc.org/2474913002
Cr-Commit-Position: refs/heads/master@{#15568}
2016-12-13 10:46:06 +00:00