Commit Graph

10445 Commits

Author SHA1 Message Date
dd460e2aa2 Fix lint errors to enable stricter PyLint rules
These fixes are needed to avoid errors after submitting
https://codereview.webrtc.org/2737963003

BUG=webrtc:7303
NOTRY=True

Review-Url: https://codereview.webrtc.org/2812273002
Cr-Commit-Position: refs/heads/master@{#17679}
2017-04-12 19:06:13 +00:00
06034c7d64 Revert of Android Logging.java: Load native library only when needed (patchset #2 id:2 of https://codereview.webrtc.org/2817593003/ )
Reason for revert:
Change in Logging.java breaking compilation (incorrect reference to enum)

Original issue's description:
> Reland of Android Logging.java: Load native library only when needed (patchset #1 id:1 of https://codereview.webrtc.org/2816753002/ )
>
> Reason for revert:
> Fix bug in original CL.
>
> Original issue's description:
> > Revert of Android Logging.java: Load native library only when needed (patchset #3 id:40001 of https://codereview.webrtc.org/2803203002/ )
> >
> > Reason for revert:
> > Breaks C++ logs in Java apps.
> >
> > Original issue's description:
> > > 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}
> > > Committed: dee5eb14e1
> >
> > TBR=sakal@webrtc.org,glaznev@webrtc.org,noahric@chromium.org,magjed@webrtc.org
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=b/36410678
> >
> > Review-Url: https://codereview.webrtc.org/2816753002
> > Cr-Commit-Position: refs/heads/master@{#17676}
> > Committed: 6e4a4427dc
>
> TBR=sakal@webrtc.org,glaznev@webrtc.org,noahric@chromium.org,brandtr@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=b/36410678
>
> Review-Url: https://codereview.webrtc.org/2817593003
> Cr-Commit-Position: refs/heads/master@{#17677}
> Committed: 297714619f

TBR=sakal@webrtc.org,glaznev@webrtc.org,noahric@chromium.org,brandtr@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=b/36410678

Review-Url: https://codereview.webrtc.org/2814133002
Cr-Commit-Position: refs/heads/master@{#17678}
2017-04-12 17:13:21 +00:00
297714619f Reland of Android Logging.java: Load native library only when needed (patchset #1 id:1 of https://codereview.webrtc.org/2816753002/ )
Reason for revert:
Fix bug in original CL.

Original issue's description:
> Revert of Android Logging.java: Load native library only when needed (patchset #3 id:40001 of https://codereview.webrtc.org/2803203002/ )
>
> Reason for revert:
> Breaks C++ logs in Java apps.
>
> Original issue's description:
> > 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}
> > Committed: dee5eb14e1
>
> TBR=sakal@webrtc.org,glaznev@webrtc.org,noahric@chromium.org,magjed@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=b/36410678
>
> Review-Url: https://codereview.webrtc.org/2816753002
> Cr-Commit-Position: refs/heads/master@{#17676}
> Committed: 6e4a4427dc

TBR=sakal@webrtc.org,glaznev@webrtc.org,noahric@chromium.org,brandtr@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=b/36410678

Review-Url: https://codereview.webrtc.org/2817593003
Cr-Commit-Position: refs/heads/master@{#17677}
2017-04-12 17:03:32 +00:00
6e4a4427dc Revert of Android Logging.java: Load native library only when needed (patchset #3 id:40001 of https://codereview.webrtc.org/2803203002/ )
Reason for revert:
Breaks C++ logs in Java apps.

Original issue's description:
> 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}
> Committed: dee5eb14e1

TBR=sakal@webrtc.org,glaznev@webrtc.org,noahric@chromium.org,magjed@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=b/36410678

Review-Url: https://codereview.webrtc.org/2816753002
Cr-Commit-Position: refs/heads/master@{#17676}
2017-04-12 16:28:24 +00:00
a79143f3e9 This CL includes the following changes:
- BUILD file reorganized, unit tests now have dedicated targets.
- "fake_polqa" is a binary producing fake output in the same format of PolqaOem64; the binary is injected for unit tests instead of the actual POLQA tool.
- Minor refactoring to inject the path to the POLQA binary instead of its parent folder.
- Unit tests for the evaluation score workers.
- Unit tests for the ApmModuleSimulator class.
- Unit tests for the test data generators: ReverberationTestDataGenerator added.

BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2811953002
Cr-Commit-Position: refs/heads/master@{#17674}
2017-04-12 13:56:25 +00:00
103ac7e7d9 AEC3 Tuning changes.
This CL adds tuning to AEC3 for the purpose of reducing the impact of
gain changes in the analog microphone gain.

BUG=chromium:710818, webrtc:6018

Review-Url: https://codereview.webrtc.org/2811283003
Cr-Commit-Position: refs/heads/master@{#17673}
2017-04-12 12:40:55 +00:00
e5fd38989d Revert of Reland "Add first part of the network_tester functionality" (patchset #3 id:40001 of https://codereview.chromium.org/2808203003/ )
Reason for revert:
Break downstream bots.

Original issue's description:
> Reland "Add first part of the network_tester functionality"
>
> BUG=webrtc:7426
>
> Review-Url: https://codereview.webrtc.org/2808203003
> Cr-Commit-Position: refs/heads/master@{#17666}
> Committed: 1c223b2f75

TBR=stefan@webrtc.org,minyue@webrtc.org,nisse@webrtc.org,terelius@webrtc.org,michaelt@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/2813193002
Cr-Commit-Position: refs/heads/master@{#17672}
2017-04-12 12:07:59 +00:00
f250100475 Add POLQA to low bandwidth audio test
BUG=webrtc:7229

Review-Url: https://codereview.webrtc.org/2804083003
Cr-Commit-Position: refs/heads/master@{#17671}
2017-04-12 12:00:56 +00:00
8d8185c774 Add command-line param to screenshare_loopback to specify a list of slides
BUG=none

Review-Url: https://codereview.webrtc.org/2814023003
Cr-Commit-Position: refs/heads/master@{#17670}
2017-04-12 11:52:55 +00:00
69ffdf4938 Further SSE2 optimizations for the AEC3 adaptive filter.
This CL adds further SSE2 optimizations for the AEC3
adaptive filter.

The changes are bitexact

BUG=webrtc:6018

Review-Url: https://codereview.webrtc.org/2810133002
Cr-Commit-Position: refs/heads/master@{#17667}
2017-04-12 10:04:09 +00:00
1c223b2f75 Reland "Add first part of the network_tester functionality"
BUG=webrtc:7426

Review-Url: https://codereview.webrtc.org/2808203003
Cr-Commit-Position: refs/heads/master@{#17666}
2017-04-12 08:50:35 +00:00
07e20db42d return comparevideos stdout and fix missing device case
BUG=webrtc:7203
NOTRY=True

Review-Url: https://codereview.webrtc.org/2809913002
Cr-Commit-Position: refs/heads/master@{#17665}
2017-04-12 08:36:02 +00:00
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