Commit Graph

120 Commits

Author SHA1 Message Date
ca4cac7e74 New module for the adaptive level controlling functionality in the audio processing module
NOTRY=true
TBR=aluebs@webrtc.org
BUG=webrtc:5920

Review-Url: https://codereview.webrtc.org/2090583002
Cr-Commit-Position: refs/heads/master@{#13333}
2016-06-29 22:26:19 +00:00
191b359d0d Implement timestamp translation/filter in VideoCapturer.
Use in AndroidVideoCapturer.

BUG=webrtc:5740

Review-Url: https://codereview.webrtc.org/2017443003
Cr-Commit-Position: refs/heads/master@{#13254}
2016-06-22 15:36:58 +00:00
3bcedd3595 GN: Add SDK tests to rtc_unittests.
In https://codereview.webrtc.org/2034923003 it was discovered
that a test binary rtc_sdk_peerconnection_objc_tests was
a dependency to rtc_unittests. Unfortunately gtest doesn't
include dependent executables into the same test executable;
only libraries (so theses tests weren't run).

This CL incorporates those tests into rtc_unittests and
does the same changes to the GN build.

BUG=webrtc:5949
TESTED=Built and ran rtc_unittests locally on Mac.
NOTRY=True

Review-Url: https://codereview.webrtc.org/2041743003
Cr-Commit-Position: refs/heads/master@{#13060}
2016-06-08 08:14:22 +00:00
f5f03e823c Reland of: Improving the fake clock and using it to fix a flaky STUN timeout test.
When the fake clock's time is advanced, it now ensures all pending
queued messages have been dispatched. This allows us to write a
"SIMULATED_WAIT" macro that ticks the simulated clock by milliseconds up
until the target time.

Useful in this case, where we know the STUN timeout should take a total
of 9500ms, but it would be overly complex to write test code that waits
for each individual timeout, ensures a STUN packet has been
retransmited, etc.

(The test described above *should* be written, but it belongs in
p2ptransportchannel_unittest.cc, not webrtcsession_unittest.cc).

Review-Url: https://codereview.webrtc.org/2024813004
Cr-Commit-Position: refs/heads/master@{#13052}
2016-06-06 18:16:13 +00:00
f83a94a41e Revert of Improving the fake clock and using it to fix a flaky STUN timeout test. (patchset #10 id:180001 of https://codereview.webrtc.org/2024813004/ )
Reason for revert:
There seems to be a TSan warning that wasn't caught by the trybot: https://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan%20v2/builds/6732/steps/peerconnection_unittests/logs/stdio

Apparently a thread is still alive even after destroying WebRTCSession. Need to think of a way to fix this, without adding a critical section around g_clock (since that would hurt performance).

Original issue's description:
> Improving the fake clock and using it to fix a flaky STUN timeout test.
>
> When the fake clock's time is advanced, it now ensures all pending
> queued messages have been dispatched. This allows us to write a
> "SIMULATED_WAIT" macro that ticks the simulated clock by milliseconds up
> until the target time.
>
> Useful in this case, where we know the STUN timeout should take a total
> of 9500ms, but it would be overly complex to write test code that waits
> for each individual timeout, ensures a STUN packet has been
> retransmited, etc.
>
> (The test described above *should* be written, but it belongs in
> p2ptransportchannel_unittest.cc, not webrtcsession_unittest.cc).
>
> Committed: https://crrev.com/ffbe0e17e2c9b7fe101023acf40574dc0c95631a
> Cr-Commit-Position: refs/heads/master@{#13043}

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

Review-Url: https://codereview.webrtc.org/2038213002
Cr-Commit-Position: refs/heads/master@{#13045}
2016-06-03 23:05:30 +00:00
ffbe0e17e2 Improving the fake clock and using it to fix a flaky STUN timeout test.
When the fake clock's time is advanced, it now ensures all pending
queued messages have been dispatched. This allows us to write a
"SIMULATED_WAIT" macro that ticks the simulated clock by milliseconds up
until the target time.

Useful in this case, where we know the STUN timeout should take a total
of 9500ms, but it would be overly complex to write test code that waits
for each individual timeout, ensures a STUN packet has been
retransmited, etc.

(The test described above *should* be written, but it belongs in
p2ptransportchannel_unittest.cc, not webrtcsession_unittest.cc).

Review-Url: https://codereview.webrtc.org/2024813004
Cr-Commit-Position: refs/heads/master@{#13043}
2016-06-03 22:31:37 +00:00
3ee5614650 GN: Add rtc_unittests
Everything except the dependency on
sdk/sdk_tests.gyp:rtc_sdk_peerconnection_objc_tests is included.
That one actually doesn't make those tests run as part of rtc_unittests
(by design). Fixing that + add them will done in aseparate CL.

BUG=webrtc:5949
TESTED=Built and ran rtc_unittests on Mac.
Verified the number of tests matched a run on the bot (1213 tests executed).

R=perkj@webrtc.org

Review URL: https://codereview.webrtc.org/2034923003 .

Cr-Commit-Position: refs/heads/master@{#13036}
2016-06-03 13:57:06 +00:00
26306b139a Remove test targets of type none.
Move the sources of rtc_unittests and xmllite_xmpp_unittests
into the actual targets instead of depending on none-targets.
This will make it easier to create GN targets matching them.

BUG=webrtc:5949
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2029583002
Cr-Commit-Position: refs/heads/master@{#13008}
2016-06-02 09:26:10 +00:00
ce17e01bf6 Reland of Android: Change camera fps range selection (patchset #1 id:1 of https://codereview.webrtc.org/2021233002/ )
Reason for revert:
Fixed gyp bug.

Original issue's description:
> Revert of Android: Change camera fps range selection (patchset #4 id:100001 of https://codereview.webrtc.org/2013413002/ )
>
> Reason for revert:
> Breaks chromium fyi:
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/13565
> on step 'generate_build_files':
> gyp: /b/build/slave/Mac_Builder/build/src/third_party/build/android/test_runner.gypi not found
>
> Original issue's description:
> > Android: Change camera fps range selection
> >
> > This CL changes the logic in
> > CameraEnumerationAndroid.getClosestSupportedFramerateRange() to prefer
> > fps ranges with a low lower bound so the camera can adjust for
> > brightness conditions.
> >
> > To test the functionality of the fps range selection, JUnit tests are
> > added. This required a new target in api_tests.gyp. JUnit tests are
> > preferable over instrumentation tests
> > (libjingle_peerconnection_android_unittest) because they are faster and
> > simpler.
> >
> > R=kjellander@webrtc.org, sakal@webrtc.org
> >
> > Committed: https://crrev.com/b4ddb5c3d3706b1c02437f6a538576f3552ab908
> > Cr-Commit-Position: refs/heads/master@{#12964}
>
> TBR=sakal@webrtc.org,kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/b3f208d0ba45f140272e3e705b5cdadc3c76514b
> Cr-Commit-Position: refs/heads/master@{#12966}

TBR=sakal@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2028583002
Cr-Commit-Position: refs/heads/master@{#12980}
2016-06-01 07:44:07 +00:00
208d19845d Rename APK tests workaround to make it more generic.
We plan to add junit tests running with Robolectric
so naming these files "apk" is slightly confusing.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2020213002
Cr-Commit-Position: refs/heads/master@{#12971}
2016-05-31 11:01:47 +00:00
b3f208d0ba Revert of Android: Change camera fps range selection (patchset #4 id:100001 of https://codereview.webrtc.org/2013413002/ )
Reason for revert:
Breaks chromium fyi:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/13565
on step 'generate_build_files':
gyp: /b/build/slave/Mac_Builder/build/src/third_party/build/android/test_runner.gypi not found

Original issue's description:
> Android: Change camera fps range selection
>
> This CL changes the logic in
> CameraEnumerationAndroid.getClosestSupportedFramerateRange() to prefer
> fps ranges with a low lower bound so the camera can adjust for
> brightness conditions.
>
> To test the functionality of the fps range selection, JUnit tests are
> added. This required a new target in api_tests.gyp. JUnit tests are
> preferable over instrumentation tests
> (libjingle_peerconnection_android_unittest) because they are faster and
> simpler.
>
> R=kjellander@webrtc.org, sakal@webrtc.org
>
> Committed: https://crrev.com/b4ddb5c3d3706b1c02437f6a538576f3552ab908
> Cr-Commit-Position: refs/heads/master@{#12964}

TBR=sakal@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2021233002
Cr-Commit-Position: refs/heads/master@{#12966}
2016-05-31 08:44:34 +00:00
b4ddb5c3d3 Android: Change camera fps range selection
This CL changes the logic in
CameraEnumerationAndroid.getClosestSupportedFramerateRange() to prefer
fps ranges with a low lower bound so the camera can adjust for
brightness conditions.

To test the functionality of the fps range selection, JUnit tests are
added. This required a new target in api_tests.gyp. JUnit tests are
preferable over instrumentation tests
(libjingle_peerconnection_android_unittest) because they are faster and
simpler.

R=kjellander@webrtc.org, sakal@webrtc.org

Review URL: https://codereview.webrtc.org/2013413002 .

Cr-Commit-Position: refs/heads/master@{#12964}
2016-05-31 08:19:02 +00:00
09c60da704 Split TaskQueue out into a separate build target.
This is needed as there are targets such as newlib_pnacl/remoting_client_plugin_newlib.pexe that depend on rtc_base_approved but don't need TaskQueue. We could implement support for TaskQueue in nacl using ppapi types, but it looks like there isn't a need for it. Libevent isn't supported for nacl either, so I'm introducing a layer on top of rtc_base_approved for TaskQueue.  It's conceivable that this target will morph into a target that holds other threading primitives such as platform_thread and possibly socket related operations, which is also an area that we currently #ifdef out for nacl in a few places.

Functional change: Removes the "is_nacl" check.

R=phoglund@webrtc.org

Review-Url: https://codereview.webrtc.org/2001913002
Cr-Commit-Position: refs/heads/master@{#12878}
2016-05-24 18:40:44 +00:00
32e80e4b2e Separate out xml dependencies and tests from rtc_unittests into a separate target.
NOTRY=true
(using notry due to offline android_arm64_rel bot)

Review-Url: https://codereview.webrtc.org/2007773003
Cr-Commit-Position: refs/heads/master@{#12869}
2016-05-24 12:48:02 +00:00
0e9d6d9e0c Add class which periodically computes statistics.
During a period (e.g. 2 seconds), different metrics can be computed, e.g.:
- average of samples
- percentage/permille of samples
- units per second

Each periodic metric can be either:
- reported to an observer each period
- aggregated during the call (e.g. min, max, average)

BUG=webrtc:5283

Review-Url: https://codereview.webrtc.org/1640053003
Cr-Commit-Position: refs/heads/master@{#12847}
2016-05-23 13:08:05 +00:00
d5c1a0bd5d New parser for event log. Manually parse the outermost EventStream to more easily deal with corrupt or partially written logs.
Changed rtpdump converter and neteq tool to use new parser, but still aborting if the file is corrupt.

Review-Url: https://codereview.webrtc.org/1768773002
Cr-Commit-Position: refs/heads/master@{#12714}
2016-05-13 07:43:04 +00:00
c6c00b32da Revert of Remove the rtc_relative_path GYP variable and similar defines (patchset #1 id:1 of https://codereview.webrtc.org/1925733002/ )
Reason for revert:
Breaks downstream gtest usage.

Original issue's description:
> Remove the rtc_relative_path GYP variable and similar defines
>
> This is a reland of https://codereview.webrtc.org/1903553003/ but with
> the SRTP changes removed, since they're needed downstream.
>
> The defines that can be used to alter the include paths for Expat and gtest
> are no longer needed in WebRTC or Chromium. Remove them to simplify GYP.
>
> Removed defines:
> EXPAT_RELATIVE_PATH
> GTEST_RELATIVE_PATH
>
> They're all set in the Chromium build so this shouldn't affect Chromium:
> https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle.gyp
>
> BUG=webrtc:4256
> NOTRY=True
> NOPRESUBMIT=True
> TBR=perkj@webrtc.org
>
> Committed: https://crrev.com/081254f2c62037d016f9fc961764c6f01cb095da
> Cr-Commit-Position: refs/heads/master@{#12536}

TBR=perkj@webrtc.org,kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:4256

Review-Url: https://codereview.webrtc.org/1945803003
Cr-Commit-Position: refs/heads/master@{#12622}
2016-05-04 08:54:39 +00:00
35151f35ec Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
- "WebRTC.Video.SendDelayInMs"

Change so that PacketOption packet id is always set in RtpSender (if having a TransportSequenceNumberAllocator).
Add SendDelayStats class for computing delays.
Add SendPacketObserver to RtpRtcp config and register SendDelayStats as observer.
Wire up OnSentPacket to SendDelayStats.

BUG=webrtc:5215

Review-Url: https://codereview.webrtc.org/1478253002
Cr-Commit-Position: refs/heads/master@{#12600}
2016-05-03 06:44:11 +00:00
b99395a544 Reland of Delete video_render module. (patchset #1 id:1 of https://codereview.webrtc.org/1923613003/ )
Reason for revert:
Chrome's build files have now been updated, see cl https://codereview.chromium.org/1929933002/

Original issue's description:
> Revert of Delete video_render module. (patchset #12 id:220001 of https://codereview.webrtc.org/1912143002/ )
>
> Reason for revert:
> This breaks every buildbot in chromium.webrtc.fyi and I don't see any roll in progress to address this (and I don't see how that would be possible either).
> Usage in Chrome: https://code.google.com/p/chromium/codesearch#search/&q=modules.gyp%3Avideo_render&sq=package:chromium&type=cs
>
> Example failures:
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/5420
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/4526
>
> I think it's fine to delete our video_render_module_internal_impl target and those files, but video_render target needs to remain.
>
> Original issue's description:
> > Delete video_render module.
> >
> > BUG=webrtc:5817
> >
> > Committed: https://crrev.com/97cfd1ec05d07ef233356e57f7aa4b028b74ffba
> > Cr-Commit-Position: refs/heads/master@{#12526}
>
> TBR=mflodman@webrtc.org,pbos@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5817

TBR=mflodman@webrtc.org,pbos@webrtc.org,kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5817

Review-Url: https://codereview.webrtc.org/1929223003
Cr-Commit-Position: refs/heads/master@{#12556}
2016-04-29 07:58:48 +00:00
081254f2c6 Remove the rtc_relative_path GYP variable and similar defines
This is a reland of https://codereview.webrtc.org/1903553003/ but with
the SRTP changes removed, since they're needed downstream.

The defines that can be used to alter the include paths for Expat and gtest
are no longer needed in WebRTC or Chromium. Remove them to simplify GYP.

Removed defines:
EXPAT_RELATIVE_PATH
GTEST_RELATIVE_PATH

They're all set in the Chromium build so this shouldn't affect Chromium:
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle.gyp

BUG=webrtc:4256
NOTRY=True
NOPRESUBMIT=True
TBR=perkj@webrtc.org

Review-Url: https://codereview.webrtc.org/1925733002
Cr-Commit-Position: refs/heads/master@{#12536}
2016-04-27 17:13:28 +00:00
0190367cea Revert of Delete video_render module. (patchset #12 id:220001 of https://codereview.webrtc.org/1912143002/ )
Reason for revert:
This breaks every buildbot in chromium.webrtc.fyi and I don't see any roll in progress to address this (and I don't see how that would be possible either).
Usage in Chrome: https://code.google.com/p/chromium/codesearch#search/&q=modules.gyp%3Avideo_render&sq=package:chromium&type=cs

Example failures:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/5420
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/4526

I think it's fine to delete our video_render_module_internal_impl target and those files, but video_render target needs to remain.

Original issue's description:
> Delete video_render module.
>
> BUG=webrtc:5817
>
> Committed: https://crrev.com/97cfd1ec05d07ef233356e57f7aa4b028b74ffba
> Cr-Commit-Position: refs/heads/master@{#12526}

TBR=mflodman@webrtc.org,pbos@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5817

Review-Url: https://codereview.webrtc.org/1923613003
Cr-Commit-Position: refs/heads/master@{#12534}
2016-04-27 15:56:56 +00:00
97cfd1ec05 Delete video_render module.
BUG=webrtc:5817

Review URL: https://codereview.webrtc.org/1912143002

Cr-Commit-Position: refs/heads/master@{#12526}
2016-04-27 09:52:27 +00:00
9eeb6240c9 Build dynamic iOS SDK.
- Places most ObjC code into webrtc/sdk/objc instead.
- New gyp targets to build, strip and export symbols for dylib.
- Removes old script used to generate dylib.

BUG=

Review URL: https://codereview.webrtc.org/1903663002

Cr-Commit-Position: refs/heads/master@{#12524}
2016-04-27 08:54:27 +00:00
4311ba59d8 Refactored CL for moving the output to a separate thread.
The logging thread is always active. The main thread uses SwapQueues to pass events to the logging thread. The logging thread moves the events to either a RingBuffer history in memory, or to a string which is written to disc.

RtcEventLogImpl constructor takes a clock for easier testing.

BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1687703002

Cr-Commit-Position: refs/heads/master@{#12476}
2016-04-22 19:40:46 +00:00
7bc7c06e6a Revert of Remove the rtc_relative_path GYP variable and similar defines (patchset #1 id:1 of https://codereview.webrtc.org/1903553003/ )
Reason for revert:
Breaks downstream for SRTP include paths. Will rework this and reland without that one.

Original issue's description:
> Remove the rtc_relative_path GYP variable and similar defines
>
> The defines that can be used to alter the include paths for Expat, SRTP
> and gtest are no longer needed in WebRTC or Chromium. Let's remove them
> to simplify the GYP a little.
>
> Removed defines:
> EXPAT_RELATIVE_PATH
> GTEST_RELATIVE_PATH
> SRTP_RELATIVE_PATH
>
> They're all set in the Chromium build so this shouldn't affect Chromium:
> https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle.gyp
>
> BUG=webrtc:4256
> NOTRY=True
> NOPRESUBMIT=True
>
> Committed: https://crrev.com/e19cf59eb6ee44fd4d7e7fbcfdd1a6ea75063605
> Cr-Commit-Position: refs/heads/master@{#12467}

TBR=pthatcher@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4256

Review URL: https://codereview.webrtc.org/1913043003

Cr-Commit-Position: refs/heads/master@{#12468}
2016-04-22 11:57:56 +00:00
e19cf59eb6 Remove the rtc_relative_path GYP variable and similar defines
The defines that can be used to alter the include paths for Expat, SRTP
and gtest are no longer needed in WebRTC or Chromium. Let's remove them
to simplify the GYP a little.

Removed defines:
EXPAT_RELATIVE_PATH
GTEST_RELATIVE_PATH
SRTP_RELATIVE_PATH

They're all set in the Chromium build so this shouldn't affect Chromium:
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle.gyp

BUG=webrtc:4256
NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1903553003

Cr-Commit-Position: refs/heads/master@{#12467}
2016-04-22 11:41:55 +00:00
e532aec252 Add isolate files for Android tests
BUG=chromium:583318
TESTED=Passing runs with:
GYP_DEFINES='test_isolation_mode=prepare OS=android' webrtc/build/gyp_webrtc
ninja -C out/Release
NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1882963003

Cr-Commit-Position: refs/heads/master@{#12397}
2016-04-18 03:08:28 +00:00
00984ff688 Reland of move {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1846693002/ )
The re-land moves the isolate build targets for media.gyp
and pc.gyp into the include_tests==1 condition.
This has been tested in a Chromium checkout and no longer
causes the error that was seen after landing
https://codereview.webrtc.org/1839763004/

Original issue's description:
> Revert of Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1839763004/ )
>
> Reason for revert:
> Breaks Chromium: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/11313/steps/gclient%20runhooks/logs/stdio:
>
> Updating projects from gyp files...
> Using overrides found in /Users/chrome-bot/.gyp/include.gypi
> Traceback (most recent call last):
>   File "src/build/gyp_chromium", line 12, in <module>
>     execfile(__file__ + '.py')
>   File "src/build/gyp_chromium.py", line 341, in <module>
>     sys.exit(main())
>   File "src/build/gyp_chromium.py", line 328, in main
>     gyp_rc = gyp.main(args)
>   File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 538, in main
>     return gyp_main(args)
>   File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 514, in gyp_main
>     options.duplicate_basename_check)
>   File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 130, in Load
>     params['parallel'], params['root_targets'])
>   File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/input.py", line 2800, in Load
>     RemoveLinkDependenciesFromNoneTargets(targets)
>   File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/input.py", line 1510, in RemoveLinkDependenciesFromNoneTargets
>     if targets[t].get('variables', {}).get('link_dependency', 0):
> KeyError: '/b/build/slave/Mac_Builder/build/src/third_party/webrtc/media/media.gyp:rtc_media_unittests#target'
> Error: Command '/usr/bin/python src/build/gyp_chromium' returned non-zero exit status 1 in /b/build/slave/Mac_Builder/build
> Hook '/usr/bin/python src/build/gyp_chromium' took 20.29 secs
>
> Original issue's description:
> > Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files.
> >
> > These contributes to circular dependency problems in WebRTC
> > since one have to depend on webrtc.gyp in order to depend on
> > a target in them.
> >
> > This reduces the number of cyclic dependencies in WebRTC from 21
> > to 16.
> >
> > BUG=webrtc:4243
> > NOTRY=True
> > NOPRESUBMIT=True
> >
> > Committed: https://crrev.com/231b69f28dd22f4e2d98e5048f8eaae7b20915e6
> > Cr-Commit-Position: refs/heads/master@{#12166}
>
> TBR=pthatcher@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4243
>
> Committed: https://crrev.com/72644d2cf6b14bbc4a107f79158eaa225f3196b5
> Cr-Commit-Position: refs/heads/master@{#12167}

TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4243

Review URL: https://codereview.webrtc.org/1843193002

Cr-Commit-Position: refs/heads/master@{#12180}
2016-03-31 14:23:52 +00:00
72644d2cf6 Revert of Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1839763004/ )
Reason for revert:
Breaks Chromium: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/11313/steps/gclient%20runhooks/logs/stdio:

Updating projects from gyp files...
Using overrides found in /Users/chrome-bot/.gyp/include.gypi
Traceback (most recent call last):
  File "src/build/gyp_chromium", line 12, in <module>
    execfile(__file__ + '.py')
  File "src/build/gyp_chromium.py", line 341, in <module>
    sys.exit(main())
  File "src/build/gyp_chromium.py", line 328, in main
    gyp_rc = gyp.main(args)
  File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 538, in main
    return gyp_main(args)
  File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 514, in gyp_main
    options.duplicate_basename_check)
  File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 130, in Load
    params['parallel'], params['root_targets'])
  File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/input.py", line 2800, in Load
    RemoveLinkDependenciesFromNoneTargets(targets)
  File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/input.py", line 1510, in RemoveLinkDependenciesFromNoneTargets
    if targets[t].get('variables', {}).get('link_dependency', 0):
KeyError: '/b/build/slave/Mac_Builder/build/src/third_party/webrtc/media/media.gyp:rtc_media_unittests#target'
Error: Command '/usr/bin/python src/build/gyp_chromium' returned non-zero exit status 1 in /b/build/slave/Mac_Builder/build
Hook '/usr/bin/python src/build/gyp_chromium' took 20.29 secs

Original issue's description:
> Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files.
>
> These contributes to circular dependency problems in WebRTC
> since one have to depend on webrtc.gyp in order to depend on
> a target in them.
>
> This reduces the number of cyclic dependencies in WebRTC from 21
> to 16.
>
> BUG=webrtc:4243
> NOTRY=True
> NOPRESUBMIT=True
>
> Committed: https://crrev.com/231b69f28dd22f4e2d98e5048f8eaae7b20915e6
> Cr-Commit-Position: refs/heads/master@{#12166}

TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4243

Review URL: https://codereview.webrtc.org/1846693002

Cr-Commit-Position: refs/heads/master@{#12167}
2016-03-30 18:54:14 +00:00
231b69f28d Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files.
These contributes to circular dependency problems in WebRTC
since one have to depend on webrtc.gyp in order to depend on
a target in them.

This reduces the number of cyclic dependencies in WebRTC from 21
to 16.

BUG=webrtc:4243
NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1839763004

Cr-Commit-Position: refs/heads/master@{#12166}
2016-03-30 18:40:38 +00:00
e0897c043b Remove webrtc/sound/ subdir.
Now that DeviceManager and DeviceInfo are gone, this code is unused.

BUG=webrtc:5579

Review URL: https://codereview.webrtc.org/1715043002

Cr-Commit-Position: refs/heads/master@{#12040}
2016-03-17 18:23:05 +00:00
0e73934694 Remove webrtc/test/webrtc_test_common.gyp
Move the "webrtc_test_common" target to test.gyp and rename
it to "test_common".

Move all tests in "webrtc_test_common_unittests" (which
wasn't run on the bots) into "test_support_unittests".

NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1754593002

Cr-Commit-Position: refs/heads/master@{#11848}
2016-03-02 18:46:25 +00:00
0206000a66 iOS: Add resource files for tests and implement OutputPath
With this change the following tests have been successfully
passing in the iOS Simulator for iPhone 5 and iOS 9:
* audio_decoder_unittests
* common_video_unittests
* modules_tests
* rtc_api_objc_tests
* rtc_pc_unittests
* system_wrappers_unittests
* voice_engine_unittests

The modules_unittests and common_audio_unittests are
handled in https://codereview.webrtc.org/1698033002/

BUG=webrtc:4755
NOTRY=True

Review URL: https://codereview.webrtc.org/1694353003

Cr-Commit-Position: refs/heads/master@{#11646}
2016-02-17 06:06:17 +00:00
e829f58928 Rename libjingle_p2p_unittest -> rtc_pc_unittests
BUG=webrtc:5419
R=pthatcher@webrtc.org
TBR=henrika@webrtc.org

Review URL: https://codereview.webrtc.org/1694813002 .

Cr-Commit-Position: refs/heads/master@{#11619}
2016-02-15 08:36:44 +00:00
efce73e4ed Rename libjingle_media_unittest -> rtc_media_unittests
Landing this must be coordinated with updating the buildbots in
https://codereview.chromium.org/1681393004

R=pthatcher@webrtc.org
TBR=henrika@webrtc.org
BUG=webrtc:5420
NOTRY=True

Review URL: https://codereview.webrtc.org/1683163002 .

Cr-Commit-Position: refs/heads/master@{#11600}
2016-02-12 14:06:11 +00:00
9b8df25c73 Move talk/session/media -> webrtc/pc
The libjingle_p2p target is renamed to rtc_pc.
The libjingle_p2p_unittest test will be renamed in a
separate follow-up CL, to make it possible to run all
trybots successfully for this CL.

BUG=webrtc:5419
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1691463002 .

Cr-Commit-Position: refs/heads/master@{#11592}
2016-02-12 05:48:10 +00:00
a96e2d77cb Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.

The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.

The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.

License headers will be updated in a follow-up CL in order to not
break Git history.

Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
  except for these files:
  talk/app/webrtc/peerconnectionendtoend_unittest.cc
  talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
  talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
  webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
  webrtc/base/testutils.cc
  webrtc/base/testutils.h

The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.

I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.

For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/

BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1587193006

Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-05 07:52:35 +00:00
ab8f82ffe0 Make ECDSA default for RTCPeerConnection
BUG=

Review URL: https://codereview.webrtc.org/1649533002

Cr-Commit-Position: refs/heads/master@{#11409}
2016-01-28 01:50:15 +00:00
bab934bffe H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding.
It works on all platforms except Android and iOS (FFmpeg limitation).

Implemented behind compile time flags, off by default.
The plan is to have it enabled in Chrome (see bug), but not in Chromium/webrtc by default.

Flags to turn it on:
- rtc_use_h264 = true
- ffmpeg_branding = "Chrome" (or other brand that includes H.264 decoder)

Tests using H264:
- video_loopback --codec=H264
- screenshare_loopback --codec=H264
- video_engine_tests (EndToEndTest.SendsAndReceivesH264)

NOTRY=True
BUG=500605, 468365
BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424

Review URL: https://codereview.webrtc.org/1306813009

Cr-Commit-Position: refs/heads/master@{#11390}
2016-01-27 09:36:07 +00:00
7b971e728b Remove extra_options from VideoCodec.
Constructing default options is racy when initializing multiple VP8
encoders in parallel. This is only used for VP8 temporal layers. Adding
TemporalLayerFactory to VP8 codec specifics instead of generic options.

Removes the last webrtc::Config uses/includes from video code.

Also removes VideoCodec equality operators which are no longer in use.

BUG=webrtc:5410
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1606613003 .

Cr-Commit-Position: refs/heads/master@{#11307}
2016-01-19 15:26:24 +00:00
ff483617a4 Step 1 to prepare call_test.* for combined audio/video tests.
Also move (and clean up includes) rampup_tests.* to webrtc/call in preparation for combined audio/video ramp-up tests.

No functional changes.

BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1537273003

Cr-Commit-Position: refs/heads/master@{#11101}
2015-12-21 11:14:05 +00:00
9fca7e18c3 A unittest that reports the statistics for the duration of an APM stream processing API call.
BUG=webrtc:5099

Committed: https://crrev.com/880896ab0976bbf86a6753d0c900c70e51f421cb
Cr-Commit-Position: refs/heads/master@{#10786}

Review URL: https://codereview.webrtc.org/1436553004

Cr-Commit-Position: refs/heads/master@{#11098}
2015-12-21 07:13:46 +00:00
aa32c3e537 Update API for Objective-C RTCIceServer
BUG=

Review URL: https://codereview.webrtc.org/1499653003

Cr-Commit-Position: refs/heads/master@{#11000}
2015-12-14 03:58:19 +00:00
7623ce4aeb Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
Reason for revert:
Bot breakage caused by TickTime::UseFakeClock has been removed.

Original issue's description:
> Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
>
> Reason for revert:
> Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots.
>
> Original issue's description:
> > Merge webrtc/video_engine/ into webrtc/video/
> >
> > BUG=webrtc:1695
> > R=mflodman@webrtc.org
> >
> > Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646
> > Cr-Commit-Position: refs/heads/master@{#10926}
>
> TBR=mflodman@webrtc.org,pbos@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:1695
>
> Committed: https://crrev.com/8237abf563bf4782ee104408b53cc8e55ce44518
> Cr-Commit-Position: refs/heads/master@{#10937}

BUG=webrtc:1695
TBR=mflodman@webrtc.org,kjellander@webrtc.org
NOPRESUBMIT=true

Review URL: https://codereview.webrtc.org/1510183002 .

Cr-Commit-Position: refs/heads/master@{#10948}
2015-12-09 11:13:40 +00:00
8237abf563 Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
Reason for revert:
Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots.

Original issue's description:
> Merge webrtc/video_engine/ into webrtc/video/
>
> BUG=webrtc:1695
> R=mflodman@webrtc.org
>
> Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646
> Cr-Commit-Position: refs/heads/master@{#10926}

TBR=mflodman@webrtc.org,pbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:1695

Review URL: https://codereview.webrtc.org/1507903005

Cr-Commit-Position: refs/heads/master@{#10937}
2015-12-08 15:12:11 +00:00
03ef053202 Merge webrtc/video_engine/ into webrtc/video/
BUG=webrtc:1695
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1506773002 .

Cr-Commit-Position: refs/heads/master@{#10926}
2015-12-08 08:09:07 +00:00
bc32ab458b Remove 'video_engine_core_unittests' binary.
Merges tests into 'video_engine_tests' to reduce the number of test
targets.

BUG=webrtc:1695
R=kjellander@webrtc.org, phoglund@webrtc.org

Review URL: https://codereview.webrtc.org/1409803007 .

Cr-Commit-Position: refs/heads/master@{#10891}
2015-12-04 09:59:02 +00:00
6e004a44e8 Revert of Created a test that reports the statistics for the duration of APM stream processing API calls. (patchset #15 id:280001 of https://codereview.webrtc.org/1436553004/ )
Reason for revert:
This breaks the Win32 Release [large tests] bot (webrtc_perf_tests times out after 1h23m): https://build.chromium.org/p/client.webrtc/builders/Win32%20Release%20%5Blarge%20tests%5D

The Mac64 Release [large tests] bot's runtime also increased with +20 minutes.

These bot configs are not a part of the default trybot set, so please run them manually or add this to the CL description:
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Original issue's description:
> A unittest that reports the statistics for the duration of an APM stream processing API call.
>
> BUG=webrtc:5099
>
> Committed: https://crrev.com/880896ab0976bbf86a6753d0c900c70e51f421cb
> Cr-Commit-Position: refs/heads/master@{#10786}

TBR=henrik.lundin@webrtc.org,solenberg@webrtc.org,peah@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5099

Review URL: https://codereview.webrtc.org/1473733004

Cr-Commit-Position: refs/heads/master@{#10791}
2015-11-25 20:27:46 +00:00
880896ab09 A unittest that reports the statistics for the duration of an APM stream processing API call.
BUG=webrtc:5099

Review URL: https://codereview.webrtc.org/1436553004

Cr-Commit-Position: refs/heads/master@{#10786}
2015-11-25 10:07:57 +00:00
0e7e259ebd Move BitrateAllocator from BitrateController logic to Call.
This is a step on the way to have variable bitrate for audio and is
intended to be as much of a no-op as possible.

BUG=webrtc:5079

Review URL: https://codereview.webrtc.org/1441673002

Cr-Commit-Position: refs/heads/master@{#10630}
2015-11-13 05:02:46 +00:00