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}
- "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}
- 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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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.orgTBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1606613003 .
Cr-Commit-Position: refs/heads/master@{#11307}
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}
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}
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}
Used for tests that cannot be run in parallel due to using non-virtual
resources such as filesystems and sockets. Initially moves socket
unittests from rtc_unittest since
PhysicalSocketTest.TestUdpReadyToSendIPv4 is one of the worst flake
offenders.
Future prospect targets are GTEST_DEATH tests that are flaky on Mac in
parallel for instance.
BUG=chromium:445880
R=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1426643003 .
Cr-Commit-Position: refs/heads/master@{#10446}
AudioSendStream will be replacing the send side of VoiceEngine channels and associated APIs. Hence, they will be used transform recorded audio into RTP/RTCP packets that can be transmitted to another party, according to given parameters.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1397123003
Cr-Commit-Position: refs/heads/master@{#10307}
The BWE expects arrival timestamps in ms, while the audio path delivered
them in us, causing the BWE to break down under the combined audio/video
BWE experiment. This was introduced in r9892 (68786d2040).
BUG=webrtc:4758
R=mflodman@webrtc.org, sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1360913004 .
Cr-Commit-Position: refs/heads/master@{#10032}
Refactoring full stack, video and screenshare tests to use the same code basis
for parametrization and initialization. This patch is done on top of recently
commited full stack graphs CL https://codereview.webrtc.org/1289933003/, but
virtually no changes have been made to full_stack_plot.py nor to the VideoAnalyzer
in full stack, except moving it to video_quality_test.cc.
Also, full_stack_samples.cc (build target) was removed and replaced with
-output_filename and -duration cmdline arguments in video_loopback and
screenshare_loopback.
The important things to review:
- video_quality_test.h
Is the structure of Params good? (examples of usage can be found in
full_stack.cc, video_loopback.cc and screenshare_loopback.cc)
- video_quality_test.cc
Is the initialization correct? The case for using Analyzer and using local
renderer are different, can they be further merged?
- webrtc_tests.gypi
Reproducing the different bitrate settings the full stack and loopback tests had
was a little bit tricky. To support both simultaneously, I added BitrateConfig
to the Params struct, as well as separate start_bitrate and target_bitrate flags
for loopback tests.
Note: Side-by-side diff for video_quality_test.cc compares that file directly
with the old full_stack.cc, so changes to VideoAnalyzer are clearly visible.
Note: Recent CL I've committed added -num_temporal_layers and -sl_discard_threshold
args to loopback tests. This was removed here. Support for streams and SVC
will be added in a CL following this one.
Review URL: https://codereview.webrtc.org/1308403003
Cr-Commit-Position: refs/heads/master@{#9969}
Updating full stack test to optionally save metadata for each frame and save it
to a file with given filename (controlled from the new full_stack_samples
executable).
Adding a Python script that reads the output generated by full stack test
and plots the graph(s).
Review URL: https://codereview.webrtc.org/1289933003
Cr-Commit-Position: refs/heads/master@{#9874}
Re-lands "Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module."
This reverts commit b933667a7f97697d6390d1eee5f378cedd9ca208.
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1259683003 .
Cr-Commit-Position: refs/heads/master@{#9661}
Placed the protobuf structures in the namespace webrtc::rtclog. Removed the message field from the DebugEvent structure, since it was not used.
Added an interface to set config information for VideoReceiveStream and VideoSendStream in the event log.
Added function to log full RTCP packets and changed RTP-logging to only log headers.
Significantly extended the unit tests for RtcEventLog.
R=ivoc@webrtc.org, minyue@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1230973005 .
Cr-Commit-Position: refs/heads/master@{#9656}
