Commit Graph

11283 Commits

Author SHA1 Message Date
844d2b9670 Reconfigure capture session in a single transaction.
If we don't reconfigure capture session in a single transaction,
RTCCameraPreviewView goes transparent when switching cameras. This is
undesired behavior.

BUG=webrtc:7177

Review-Url: https://codereview.webrtc.org/2811643006
Cr-Commit-Position: refs/heads/master@{#17664}
2017-04-12 08:27:44 +00:00
5e79b29313 Adding new functionality for SIMD optimizations in AEC3
Most of the complex functionality in AEC3 is done using
vector maths. This CL adds a new functionality for
performing these using SIMD operations in a simple manner
whenever such are available.

The reason for putting the implementations in the header file
is to allow any possible inlining.

BUG=webrtc:6018

Review-Url: https://codereview.webrtc.org/2813823002
Cr-Commit-Position: refs/heads/master@{#17663}
2017-04-12 08:20:45 +00:00
0426222f4c Modified the rtp_receiver_unittests.
Implemented operator == in RtpSource and use the gmock EXPECT_THAT to make the test cleaner.

Related CL: https://codereview.webrtc.org/2770233003/

BUG=chromium:703122

Review-Url: https://codereview.webrtc.org/2813753002
Cr-Commit-Position: refs/heads/master@{#17659}
2017-04-11 18:28:10 +00:00
00d802b6ee Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
Reason for revert:
Fix failing bots.

BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2816493002
Cr-Commit-Position: refs/heads/master@{#17658}
2017-04-11 17:34:31 +00:00
10d095d4f7 Revert of Change NetEq::InsertPacket to take an RTPHeader (patchset #2 id:20001 of https://codereview.webrtc.org/2807273004/ )
Reason for revert:
Broke downstream dependencies.

Original issue's description:
> Change NetEq::InsertPacket to take an RTPHeader
>
> It used to take a WebRtcRTPHeader as input, which has an RTPHeader as
> a member. None of the other member in WebRtcRTPHeader where used in
> NetEq.
>
> This CL adapts the production code; tests and tools will be converted
> in a follow-up CL.
>
> BUG=webrtc:7467
>
> Review-Url: https://codereview.webrtc.org/2807273004
> Cr-Commit-Position: refs/heads/master@{#17652}
> Committed: 4d027576a6

TBR=ivoc@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7467

Review-Url: https://codereview.webrtc.org/2812933002
Cr-Commit-Position: refs/heads/master@{#17657}
2017-04-11 14:47:59 +00:00
80ff00cd2b Improve USB device reset logic
BUG=webrtc:7203
NOTRY=True

Review-Url: https://codereview.webrtc.org/2789533002
Cr-Commit-Position: refs/heads/master@{#17656}
2017-04-11 14:40:26 +00:00
b213a16b28 Finalized the SSE2 optimizations for the matched filter in AEC3
The SSE2 optimizations of the filter core in the matched
filter was only half-done. This CL finalizes those.

In particular:
-It adds finalization of updating of the filter.
-It removes the manual loop unrolling in order to reduce and
simplify the code.

Note that the changes pass the bitexactness tests in an
external AEC3 test suite, and the test
MatchedFilter.TestOptimizations succeed.

BUG=webrtc:6018

Review-Url: https://codereview.webrtc.org/2813563003
Cr-Commit-Position: refs/heads/master@{#17655}
2017-04-11 14:12:29 +00:00
27c46e2872 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #4 id:400001 of https://codereview.webrtc.org/2812913002/ )
Reason for revert:
Breaks android buildbots.

Original issue's description:
> Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
>
> Reason for revert:
> Reland with appropriate changes to API to not break depending projects.
>
> Original issue's description:
> > Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
> >
> > Reason for revert:
> > Breaks dependent projects.
> >
> > Original issue's description:
> > > Add content type information to Encoded Images and add corresponding RTP extension header.
> > > Use it to separate UMA e2e delay metric between screenshare from video.
> > > Content type extension is set based on encoder settings and processed and decoders.
> > >
> > > Also,
> > > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> > >
> > > BUG=webrtc:7420
> > >
> > > Review-Url: https://codereview.webrtc.org/2772033002
> > > Cr-Commit-Position: refs/heads/master@{#17640}
> > > Committed: 64e739aeae
> >
> > TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2816463002
> > Cr-Commit-Position: refs/heads/master@{#17644}
> > Committed: 5721866808
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2812913002
> Cr-Commit-Position: refs/heads/master@{#17651}
> Committed: 774f6b4b96

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2809653004
Cr-Commit-Position: refs/heads/master@{#17653}
2017-04-11 13:20:05 +00:00
4d027576a6 Change NetEq::InsertPacket to take an RTPHeader
It used to take a WebRtcRTPHeader as input, which has an RTPHeader as
a member. None of the other member in WebRtcRTPHeader where used in
NetEq.

This CL adapts the production code; tests and tools will be converted
in a follow-up CL.

BUG=webrtc:7467

Review-Url: https://codereview.webrtc.org/2807273004
Cr-Commit-Position: refs/heads/master@{#17652}
2017-04-11 13:17:46 +00:00
774f6b4b96 Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
Reason for revert:
Reland with appropriate changes to API to not break depending projects.

Original issue's description:
> Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
>
> Reason for revert:
> Breaks dependent projects.
>
> Original issue's description:
> > Add content type information to Encoded Images and add corresponding RTP extension header.
> > Use it to separate UMA e2e delay metric between screenshare from video.
> > Content type extension is set based on encoder settings and processed and decoders.
> >
> > Also,
> > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> >
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2772033002
> > Cr-Commit-Position: refs/heads/master@{#17640}
> > Committed: 64e739aeae
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2816463002
> Cr-Commit-Position: refs/heads/master@{#17644}
> Committed: 5721866808

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2812913002
Cr-Commit-Position: refs/heads/master@{#17651}
2017-04-11 13:12:37 +00:00
268862c5e4 Address denicija's comments for AppRTCMobile video codec setting.
Comments in review: https://codereview.webrtc.org/2735303004/

BUG=webrtc:7316

Review-Url: https://codereview.webrtc.org/2807533004
Cr-Commit-Position: refs/heads/master@{#17650}
2017-04-11 12:36:43 +00:00
24da37b0bf ObjC: RTCVideoSource cleanup
RTCVideoSource was recently added in
https://codereview.webrtc.org/2745193002/. This CL addresses some post
commit feedback.

BUG=webrtc:7177

Review-Url: https://codereview.webrtc.org/2812533003
Cr-Commit-Position: refs/heads/master@{#17649}
2017-04-11 11:50:15 +00:00
29dbb1992a Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2811963002/ )
Reason for revert:
Relanded by mistake.

Original issue's description:
> Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
>
> Reason for revert:
> Reland with fixes which break API
>
> Original issue's description:
> > Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
> >
> > Reason for revert:
> > Breaks dependent projects.
> >
> > Original issue's description:
> > > Add content type information to Encoded Images and add corresponding RTP extension header.
> > > Use it to separate UMA e2e delay metric between screenshare from video.
> > > Content type extension is set based on encoder settings and processed and decoders.
> > >
> > > Also,
> > > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> > >
> > > BUG=webrtc:7420
> > >
> > > Review-Url: https://codereview.webrtc.org/2772033002
> > > Cr-Commit-Position: refs/heads/master@{#17640}
> > > Committed: 64e739aeae
> >
> > TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2816463002
> > Cr-Commit-Position: refs/heads/master@{#17644}
> > Committed: 5721866808
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2811963002
> Cr-Commit-Position: refs/heads/master@{#17645}
> Committed: 4fa0c4f97f

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2810923004
Cr-Commit-Position: refs/heads/master@{#17648}
2017-04-11 11:49:07 +00:00
dee5eb14e1 Android Logging.java: Load native library only when needed
Logging.java currently always tries to load jingle_peerconnection_so in
the static section, but some clients don't want to use it. This CL loads
jingle_peerconnection_so only when a client requests it by calling one
of:
 * Logging.enableLogThreads
 * Logging.enableLogTimeStamps
 * Logging.enableTracing
 * Logging.enableLogToDebugOutput

BUG=b/36410678

Review-Url: https://codereview.webrtc.org/2803203002
Cr-Commit-Position: refs/heads/master@{#17647}
2017-04-11 11:21:50 +00:00
4fa0c4f97f Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
Reason for revert:
Reland with fixes which break API

Original issue's description:
> Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
>
> Reason for revert:
> Breaks dependent projects.
>
> Original issue's description:
> > Add content type information to Encoded Images and add corresponding RTP extension header.
> > Use it to separate UMA e2e delay metric between screenshare from video.
> > Content type extension is set based on encoder settings and processed and decoders.
> >
> > Also,
> > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> >
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2772033002
> > Cr-Commit-Position: refs/heads/master@{#17640}
> > Committed: 64e739aeae
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2816463002
> Cr-Commit-Position: refs/heads/master@{#17644}
> Committed: 5721866808

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2811963002
Cr-Commit-Position: refs/heads/master@{#17645}
2017-04-11 11:01:43 +00:00
5721866808 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
Reason for revert:
Breaks dependent projects.

Original issue's description:
> Add content type information to Encoded Images and add corresponding RTP extension header.
> Use it to separate UMA e2e delay metric between screenshare from video.
> Content type extension is set based on encoder settings and processed and decoders.
>
> Also,
> Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
>
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2772033002
> Cr-Commit-Position: refs/heads/master@{#17640}
> Committed: 64e739aeae

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2816463002
Cr-Commit-Position: refs/heads/master@{#17644}
2017-04-11 10:59:43 +00:00
cde2528d28 Enabling 'gn check' on //webrtc/ortc.
BUG=webrtc:6828

Review-Url: https://codereview.webrtc.org/2804663002
Cr-Commit-Position: refs/heads/master@{#17642}
2017-04-11 09:52:49 +00:00
10fc0e6385 Delay based logging.
BUG=none

Review-Url: https://codereview.webrtc.org/2808833002
Cr-Commit-Position: refs/heads/master@{#17641}
2017-04-11 08:50:23 +00:00
64e739aeae Add content type information to Encoded Images and add corresponding RTP extension header.
Use it to separate UMA e2e delay metric between screenshare from video.
Content type extension is set based on encoder settings and processed and decoders.

Also,
Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.

BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2772033002
Cr-Commit-Position: refs/heads/master@{#17640}
2017-04-11 08:46:04 +00:00
93cda2ebde APM-QA tool, renaming noise generators into input-reference generators.
This CL changes the name of classes, methods and variables making using "noise generator".
This naming is replaced with "input-reference generator" which is more descriptive of the actual role.
Comments, CSS class and HTML item names have also been changed.
Consistency for variable names has been verified and the style checked with pylint.

BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2805653002
Cr-Commit-Position: refs/heads/master@{#17639}
2017-04-11 08:06:28 +00:00
9765370416 Resolve dependency between rtc_event_log_api and remote_bitrate_estimator
BUG=webrtc:7257

Review-Url: https://codereview.webrtc.org/2800633004
Cr-Commit-Position: refs/heads/master@{#17638}
2017-04-11 07:49:44 +00:00
7fb7bbd179 Revert of Add first part of the network_tester functionality. (patchset #13 id:260001 of https://codereview.webrtc.org/2779233002/ )
Reason for revert:
Tasn test failure.

Original issue's description:
> Add first part of the network_tester functionality.
>
> BUG=webrtc:7426
>
> Review-Url: https://codereview.webrtc.org/2779233002
> Cr-Commit-Position: refs/heads/master@{#17635}
> Committed: 333d0ff631

TBR=stefan@webrtc.org,minyue@webrtc.org,nisse@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7426

Review-Url: https://codereview.webrtc.org/2800403003
Cr-Commit-Position: refs/heads/master@{#17636}
2017-04-11 07:16:51 +00:00
333d0ff631 Add first part of the network_tester functionality.
BUG=webrtc:7426

Review-Url: https://codereview.webrtc.org/2779233002
Cr-Commit-Position: refs/heads/master@{#17635}
2017-04-11 06:26:35 +00:00
e0ab0ad85d Rename COMPILE_ASSERT macro to RTC_COMPILE_ASSERT
This is needed to avoid name collision with some downstream projects.

BUG=b/37224347
TBR=henrika@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2808343002
Cr-Commit-Position: refs/heads/master@{#17634}
2017-04-11 06:21:43 +00:00
0d4e068d0a Make safe_cmp::* constexpr
Because it's easy and generally useful, and because a later CL in this
series needs it.

BUG=webrtc:7459

Review-Url: https://codereview.webrtc.org/2808603002
Cr-Commit-Position: refs/heads/master@{#17633}
2017-04-11 05:44:07 +00:00
8c459f9eee Restore old (deprecated) signature of initializeAndroidGlobals.
This CL removed a couple parameters from the method, and changed the
type of the first parameter to an android.content.Context:
https://codereview.webrtc.org/2800353002/

But applications still using the old method may have already upcast the
context parameter to an Object, in which case this is a breaking change.

So, leaving the old signature exactly as it was before, for maximum
backwards compatibility.

BUG=webrtc:3416
TBR=magjed@webrtc.org

Review-Url: https://codereview.webrtc.org/2810973002
Cr-Commit-Position: refs/heads/master@{#17630}
2017-04-11 01:07:55 +00:00
20c84ccd48 Making FakeNetworkPipe demux audio and video packets.
BUG=None

Review-Url: https://codereview.webrtc.org/2794243002
Cr-Commit-Position: refs/heads/master@{#17629}
2017-04-10 23:57:57 +00:00
d9ce76444f Make RtpTransport actually implement RtpTransportInterface
BUG=webrtc:7013

Review-Url: https://codereview.webrtc.org/2805783002
Cr-Commit-Position: refs/heads/master@{#17628}
2017-04-10 23:17:57 +00:00
b4fc73a3ab Removing unnecessary parameters from initializeAndroidGlobals.
The "initialize audio/video" parameters are no longer needed, but
at the same time were required to be true, causing a lot of confusion.
This CL removes them, but leaves the old method signature around,
marked "deprecated".

BUG=webrtc:3416
TBR=solenberg@webrtc.org

Review-Url: https://codereview.webrtc.org/2800353002
Cr-Commit-Position: refs/heads/master@{#17626}
2017-04-10 22:08:02 +00:00
6799553a2c Add information about microphone gain changes to AEC3
Changes in the microphone gain are effecting the AEC in the sense
that each change in the microphone gain is a change in the echo
path seen by the AEC. This CL utilizes the ability of AEC3 to
leverage information about known changes in the analog microphone
gain.

BUG=webrtc:6018

Review-Url: https://codereview.webrtc.org/2808073002
Cr-Commit-Position: refs/heads/master@{#17625}
2017-04-10 21:12:41 +00:00
6d822adac4 Added forced zero AEC output after call startup and echo path changes
During the first few capture frames, there is no way for the AEC
to tell whether there is echo in the capture signal as the echo
removal functionality in the AEC has not yet seen any render
signal. To avoid initial echo bursts due to this, this CL adds
functionality for forcing the echo suppression gain to zero during
the first 50 blocks (200 ms) after call start and after a reported
echo path change.

BUG=webrtc:6018

Review-Url: https://codereview.webrtc.org/2808733002
Cr-Commit-Position: refs/heads/master@{#17624}
2017-04-10 20:52:14 +00:00
ca31f175e1 Remove deprecated RTPPayloadStrategy
Remove deprecated set_use_rtx_payload_mapping_on_restore()
Remove unused headers

BUG=None

Review-Url: https://codereview.webrtc.org/2808743002
Cr-Commit-Position: refs/heads/master@{#17623}
2017-04-10 15:45:29 +00:00
a1ef71f622 Add parser to visualise the ana dump
BUG=webrtc:7160

Review-Url: https://codereview.webrtc.org/2696133003
Cr-Commit-Position: refs/heads/master@{#17622}
2017-04-10 15:31:26 +00:00
8d609f6b6d Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
Reason for revert:
Re-land, reverting did not fix bug.

https://bugs.chromium.org/p/webrtc/issues/detail?id=7465

Original issue's description:
> Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
>
> Reason for revert:
> Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
>
> https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
>
> Original issue's description:
> > Added the GetSources() to the RtpReceiverInterface and implemented
> > it for the AudioRtpReceiver.
> >
> > This method returns a vector of RtpSource(both CSRC source and SSRC
> > source) which contains the ID of a source, the timestamp, the source
> > type (SSRC or CSRC) and the audio level.
> >
> > The RtpSource objects are buffered and maintained by the
> > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> > the info of the contributing source will be pulled along the object
> > chain:
> > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> > AudioReceiveStream -> voe::Channel -> RtpRtcp module
> >
> > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
> >
> > BUG=chromium:703122
> > TBR=stefan@webrtc.org, danilchap@webrtc.org
> >
> > Review-Url: https://codereview.webrtc.org/2770233003
> > Cr-Commit-Position: refs/heads/master@{#17591}
> > Committed: 292084c376
>
> TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:703122
>
> Review-Url: https://codereview.webrtc.org/2809613002
> Cr-Commit-Position: refs/heads/master@{#17616}
> Committed: fbcc5cb386

TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:703122

Review-Url: https://codereview.webrtc.org/2810623003
Cr-Commit-Position: refs/heads/master@{#17621}
2017-04-10 14:39:05 +00:00
b0f7e39fd4 Move IsIntlike to type_traits.h
I'll start using it outside safe_compare.h soon.

BUG=webrtc:7459

Review-Url: https://codereview.webrtc.org/2809513002
Cr-Commit-Position: refs/heads/master@{#17620}
2017-04-10 13:56:58 +00:00
37e99fd3fa Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/
AudioDecoder and AudioDecoderFactory are in webrtc/api/ now, so move
their mocks to someplace central where tests from all over WebRTC are
allowed to #include them.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2798063004
Cr-Commit-Position: refs/heads/master@{#17619}
2017-04-10 12:15:48 +00:00
fbcc5cb386 Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
Reason for revert:
Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see

https://bugs.chromium.org/p/webrtc/issues/detail?id=7465

Original issue's description:
> Added the GetSources() to the RtpReceiverInterface and implemented
> it for the AudioRtpReceiver.
>
> This method returns a vector of RtpSource(both CSRC source and SSRC
> source) which contains the ID of a source, the timestamp, the source
> type (SSRC or CSRC) and the audio level.
>
> The RtpSource objects are buffered and maintained by the
> RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> the info of the contributing source will be pulled along the object
> chain:
> AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> AudioReceiveStream -> voe::Channel -> RtpRtcp module
>
> Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
>
> BUG=chromium:703122
> TBR=stefan@webrtc.org, danilchap@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2770233003
> Cr-Commit-Position: refs/heads/master@{#17591}
> Committed: 292084c376

TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:703122

Review-Url: https://codereview.webrtc.org/2809613002
Cr-Commit-Position: refs/heads/master@{#17616}
2017-04-10 11:38:13 +00:00
925e9d762c Removed workaround for the WARN_UNUSED_RESULT issue.
BUG=webrtc:6018

Review-Url: https://codereview.webrtc.org/2810533003
Cr-Commit-Position: refs/heads/master@{#17615}
2017-04-10 11:18:38 +00:00
4fb651dd22 Event log cleanup in tests.
TBR=stefan@webrtc.org
BUG=none

Review-Url: https://codereview.webrtc.org/2806723002
Cr-Commit-Position: refs/heads/master@{#17614}
2017-04-10 10:54:05 +00:00
fca900aa37 Fix two invalid DCHECKs related to audio BWE.
These are invalid since:
- An allocated bitrate of 0 means that the stream should be disabled. Changing the behavior to send audio at min bitrate even though the allocator asks for the stream to be disabled.
- It should be OK to set a min bitrate equal to the start bitrate.

BUG=webrtc:5079

Review-Url: https://codereview.webrtc.org/2806163003
Cr-Commit-Position: refs/heads/master@{#17613}
2017-04-10 10:53:00 +00:00
49cad02cf3 Ignore some UBSan errors
They proved to be too difficult to fix properly, so we revert the
saturation fixes that were done in
https://codereview.webrtc.org/2729573002 and
https://codereview.webrtc.org/2746903002, and ignore them instead.

BUG=webrtc:7307, chromium:709364, chromium:693868

Review-Url: https://codereview.webrtc.org/2809483002
Cr-Commit-Position: refs/heads/master@{#17612}
2017-04-10 09:29:33 +00:00
9f2c18e237 Changed OLA window for neteq. Old code didnt work well with 48khz
fixing white spaces

updated authors file

Changed OLA window to use Q14 as Q5 dosnt work with 48khz. 1 ms @ 48 khz is > 2^5

BUG=webrtc:1361

Review-Url: https://codereview.webrtc.org/2763273003
Cr-Commit-Position: refs/heads/master@{#17611}
2017-04-10 09:22:46 +00:00
c547e84ec5 Allow rtp::Packet::*RawExtension to take 0 as an extension id
BUG=webrtc:7433

Review-Url: https://codereview.webrtc.org/2803623004
Cr-Commit-Position: refs/heads/master@{#17610}
2017-04-10 08:31:49 +00:00
02465b8a11 Add some unit tests to vie_encoder.
BUG=none

Review-Url: https://codereview.webrtc.org/2801293002
Cr-Commit-Position: refs/heads/master@{#17609}
2017-04-10 08:12:52 +00:00
36e6a8f1bd WavReaderAdaptor is a simple adaptor of the existing class WavReader from webrtc/common_audio/wav_file.h. The adaptor was mainly needed to use dependency injection and easily test the MultiEndCall class (see https://codereview.webrtc.org/2761853002/).
The unit test ConversationalSpeechTest.MultiEndCallWavReaderAdaptorSine uses CreateSineWavFile() and writes temporary wav files that are used for test (deleted only if the test passes).

BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2774423005
Cr-Commit-Position: refs/heads/master@{#17608}
2017-04-10 07:53:53 +00:00
2042c16be0 Revert of Delete class ScopedPtrCollection. Replaced with vector of unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2808463002/ )
Reason for revert:
Deleting scopedptrcollection.h broke an internal project.

Original issue's description:
> Delete class ScopedPtrCollection. Replaced with vector of unique_ptr.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2808463002
> Cr-Commit-Position: refs/heads/master@{#17605}
> Committed: 188596f20f

TBR=pthatcher@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None

Review-Url: https://codereview.webrtc.org/2812553002
Cr-Commit-Position: refs/heads/master@{#17607}
2017-04-10 07:31:33 +00:00
188596f20f Delete class ScopedPtrCollection. Replaced with vector of unique_ptr.
BUG=None

Review-Url: https://codereview.webrtc.org/2808463002
Cr-Commit-Position: refs/heads/master@{#17605}
2017-04-10 07:02:52 +00:00
4b37127414 Fix compilation issues of std::unique_ptr
This patch fixes compilation issues related to usage of std::unique_ptr
and NULL instead of nullptr. This issue pops up once you would try to
compile whole webrtc with using C++14 and gcc-4.9

BUG=webrtc:7461

Review-Url: https://codereview.webrtc.org/2806693004
Cr-Commit-Position: refs/heads/master@{#17600}
2017-04-09 16:09:06 +00:00
66e9f7630f Adjust parameter in vp9 videoprocessor_integration test.
Needed for libvpx roll, to prevent failure on arm.

TBR=marpan@webrtc.org
BUG=None

Review-Url: https://codereview.webrtc.org/2804413002
Cr-Commit-Position: refs/heads/master@{#17593}
2017-04-07 22:07:18 +00:00
8d23c050f2 MultiEndCall::CheckTiming() verifies that a set of audio tracks and timing information is valid to simulate conversational speech. Unordered turns are rejected. Self cross-talk and cross-talk with 3 or more speakers are not permitted since it would require mixing at the simulation step.
This CL includes extensive tests to match accept or reject decisions on several different timing setups. The setups are simulated using mocks (by far more light-weight than using actual timing and audio track files).

The client code, the unit tests in this case, passes information about the fake audio tracks to MockWavReaderFactory. MockWavReader instances are then created using the parameters defined in the client code. To improve the readability of the tests, generator_unittest.cc includes a docstring explaining how each MultiEndCallSetup* test is documented.

Run tests as follows:
$ out/Default/modules_unittests --gtest_filter=ConversationalSpeechTest.*

BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2781573002
Cr-Commit-Position: refs/heads/master@{#17592}
2017-04-07 19:05:08 +00:00