Move talk/media to webrtc/media

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

Committed: https://crrev.com/a96e2d77cb80174af76fa3e60d7c98f6ab565f57
Cr-Commit-Position: refs/heads/master@{#11495}

[kjellander_webrtc, 2016-01-18 08:00:54]

Patchset #2 (id:10001) has been deleted

[kjellander_webrtc, 2016-01-18 08:01:09]

Patchset #2 (id:20001) has been deleted

[kjellander_webrtc, 2016-01-18 13:17:27]

Patchset #3 (id:40001) has been deleted

[kjellander_webrtc, 2016-01-18 14:25:09]

Description was changed from

==========
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.

Other modifications:
* Updated the header guards with tools/refactoring/fixincludeguards.py
* Sorted the includes using chromium/src/tools/sort-headers.py
  (one exception: talk/app/webrtc/peerconnectionendtoend_unittest.cc
   doens't compile unless one header is restored)
* Unused GYP reference to libjingle_tests_additional_deps was removed.

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.

BUG=webrtc:5420
==========

to

==========
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.

Other modifications:
* Updated the header guards with tools/refactoring/fixincludeguards.py
* Sorted the includes using chromium/src/tools/sort-headers.py
  with two exceptions: talk/app/webrtc/peerconnectionendtoend_unittest.cc
  and webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.

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.

BUG=webrtc:5420
==========

[kjellander_webrtc, 2016-01-18 16:26:08]

Patchset #4 (id:60001) has been deleted

[kjellander_webrtc, 2016-01-18 17:08:37]

Description was changed from

==========
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.

Other modifications:
* Updated the header guards with tools/refactoring/fixincludeguards.py
* Sorted the includes using chromium/src/tools/sort-headers.py
  with two exceptions: talk/app/webrtc/peerconnectionendtoend_unittest.cc
  and webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.

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.

BUG=webrtc:5420
==========

to

==========
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.

Other modifications:
* Updated the header guards with tools/refactoring/fixincludeguards.py
* 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.

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.

BUG=webrtc:5420
==========

[kjellander_webrtc, 2016-01-18 17:08:38]

kjellander@webrtc.org changed reviewers:
+ pthatcher@webrtc.org

[kjellander_webrtc, 2016-01-18 18:38:35]

Description was changed from

==========
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.

Other modifications:
* Updated the header guards with tools/refactoring/fixincludeguards.py
* 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.

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.

BUG=webrtc:5420
==========

to

==========
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.

Other modifications:
* Updated the header guards with tools/refactoring/fixincludeguards.py
* 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.

BUG=webrtc:5420
==========

[kjellander_webrtc, 2016-01-18 19:45:27]

The CQ bit was checked by kjellander@webrtc.org to run a CQ dry run

[commit-bot: I haz the power, 2016-01-18 19:45:42]

Dry run: CQ is trying da patch. Follow status at
 https://chromium-cq-status.appspot.com/patch-status/1587193006/80001
View timeline at
 https://chromium-cq-status.appspot.com/patch-timeline/1587193006/80001

[commit-bot: I haz the power, 2016-01-18 19:56:42]

The CQ bit was unchecked by commit-bot@chromium.org

[commit-bot: I haz the power, 2016-01-18 19:56:44]

Dry run: Try jobs failed on following builders:
  presubmit on tryserver.webrtc (JOB_FAILED,
http://build.chromium.org/p/tryserver.webrtc/builders/presubmit/builds/2921)

[kjellander_webrtc, 2016-01-18 20:52:37]

The CQ bit was checked by kjellander@webrtc.org to run a CQ dry run

[commit-bot: I haz the power, 2016-01-18 20:52:52]

Dry run: CQ is trying da patch. Follow status at
 https://chromium-cq-status.appspot.com/patch-status/1587193006/90001
View timeline at
 https://chromium-cq-status.appspot.com/patch-timeline/1587193006/90001

[commit-bot: I haz the power, 2016-01-18 21:02:06]

The CQ bit was unchecked by commit-bot@chromium.org

[commit-bot: I haz the power, 2016-01-18 21:02:07]

Dry run: Try jobs failed on following builders:
  presubmit on tryserver.webrtc (JOB_FAILED,
http://build.chromium.org/p/tryserver.webrtc/builders/presubmit/builds/2924)

[kjellander_webrtc, 2016-01-19 07:05:48]

kjellander@webrtc.org changed reviewers:
+ tommi@webrtc.org

[kjellander_webrtc, 2016-01-19 07:05:50]

pthatcher: main review.
tommi: feel free to review, but mostly owners rubberstamp.

[kjellander_webrtc, 2016-01-19 21:03:20]

On 2016/01/19 07:05:50, kjellander (webrtc) wrote:
> pthatcher: main review.
> tommi: feel free to review, but mostly owners rubberstamp.

I realize I haven't figured out exactly what is needed to support Chromium in
here, I guess build_with_chromium conditions and also a BUILD.gn file, so feel
free to hold off review for now.

[kjellander_webrtc, 2016-01-20 15:08:13]

On 2016/01/19 21:03:20, kjellander (webrtc) wrote:
> On 2016/01/19 07:05:50, kjellander (webrtc) wrote:
> > pthatcher: main review.
> > tommi: feel free to review, but mostly owners rubberstamp.
> 
> I realize I haven't figured out exactly what is needed to support Chromium in
> here, I guess build_with_chromium conditions and also a BUILD.gn file, so feel
> free to hold off review for now.

Update: I have now looked at the required Chromium changes and luckily they seem
very simple for talk/media (just changed locations). I have prepared
https://codereview.chromium.org/1604303002/ for that.
Feel free to review this now, every day that passes calls for a new rebase :(

[kjellander_webrtc, 2016-01-20 15:45:03]

Patchset #7 (id:100001) has been deleted

[kjellander_webrtc, 2016-01-20 16:05:08]

Patchset #7 (id:110001) has been deleted

[kjellander_webrtc, 2016-01-20 16:10:22]

Description was changed from

==========
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.

Other modifications:
* Updated the header guards with tools/refactoring/fixincludeguards.py
* 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.

BUG=webrtc:5420
==========

to

==========
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.

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.

BUG=webrtc:5420
==========

[kjellander_webrtc, 2016-01-20 17:45:45]

Patchset #7 (id:120001) has been deleted

[pbos-webrtc, 2016-01-20 18:40:49]

On 2016/01/20 15:08:13, kjellander (webrtc) wrote:
> On 2016/01/19 21:03:20, kjellander (webrtc) wrote:
> > On 2016/01/19 07:05:50, kjellander (webrtc) wrote:
> > > pthatcher: main review.
> > > tommi: feel free to review, but mostly owners rubberstamp.
> > 
> > I realize I haven't figured out exactly what is needed to support Chromium
in
> > here, I guess build_with_chromium conditions and also a BUILD.gn file, so
feel
> > free to hold off review for now.
> 
> Update: I have now looked at the required Chromium changes and luckily they
seem
> very simple for talk/media (just changed locations). I have prepared
>
https://codereview.chromium.org/1604303002/?_ga=1.153186300.532910034.1361177937
for that.
> Feel free to review this now, every day that passes calls for a new rebase :(

FYI pthatcher@ is out this week.

[kjellander_webrtc, 2016-01-20 21:11:47]

Description was changed from

==========
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.

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.

BUG=webrtc:5420
==========

to

==========
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.

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
==========

[kjellander_webrtc, 2016-01-20 21:12:15]

kjellander@webrtc.org changed reviewers:
+ deadbeef@webrtc.org

[kjellander_webrtc, 2016-01-20 21:12:18]

Since Peter is out this week: Taylor, can you have a look to see if you can spot
anything that may need to be updated.

https://codereview.webrtc.org/1587193006/diff/90001/talk/build/common.gypi
File talk/build/common.gypi (right):

https://codereview.webrtc.org/1587193006/diff/90001/talk/build/common.gypi#ne...
talk/build/common.gypi:137: 'HASH_NAMESPACE=__gnu_cxx',
I forgot to ask about these, _DEBUG and WEBRTC_POSIX are similarly defined in
the webrtc/build/common.gypi but HASH_NAMESPACE=__gnu_cxx', DISABLE_DYNAMIC_CAST
and _REENTRANT are missing.

Should I set them at the targets for the moved code just to be sure? I can't
find any references to them in the code itself, but I don't know if they're
referenced in system headers that the code may be using.
Previous moving of similar talk/ code (into webrtc/p2p and webrtc/libjingle/*
didn't add anything like that).

https://codereview.webrtc.org/1587193006/diff/90001/webrtc/api/api.gyp
File webrtc/api/api.gyp (right):

https://codereview.webrtc.org/1587193006/diff/90001/webrtc/api/api.gyp#newcode66
webrtc/api/api.gyp:66: # TODO(kjellander): Make the code compile without
disabling these.
Without this I get:
../../webrtc/api/objc/RTCSessionDescription.mm:90:1: error: control may reach
end of non-void function [-Werror,-Wreturn-type]
}
^

I can't explain why this is needed now but not before, since this code hasn't
moved nor any of the headers it includes.

[kjellander_webrtc, 2016-01-20 21:13:24]

The CQ bit was checked by kjellander@webrtc.org to run a CQ dry run

[commit-bot: I haz the power, 2016-01-20 21:13:36]

Dry run: CQ is trying da patch. Follow status at
 https://chromium-cq-status.appspot.com/patch-status/1587193006/130001
View timeline at
 https://chromium-cq-status.appspot.com/patch-timeline/1587193006/130001

[commit-bot: I haz the power, 2016-01-20 21:18:14]

The CQ bit was unchecked by commit-bot@chromium.org

[commit-bot: I haz the power, 2016-01-20 21:18:15]

Dry run: Try jobs failed on following builders:
  ios64_sim_dbg on tryserver.webrtc (JOB_FAILED,
http://build.chromium.org/p/tryserver.webrtc/builders/ios64_sim_dbg/builds/4634)

[kjellander_webrtc, 2016-01-22 08:48:46]

Description was changed from

==========
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.

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
==========

to

==========
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/.

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
==========

[pthatcher1, 2016-01-27 01:34:09]

This looks so good. 

Most of my comments are "nice to have"s.  The biggest one is just if we can get
more files to show up as moves instead of adds.

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/base/crypto...
File webrtc/media/base/cryptoparams.h (right):

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/base/crypto...
webrtc/media/base/cryptoparams.h:16: namespace cricket {
Some of these files show up as adds instead of moves.  Any idea why?

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/base/mediac...
File webrtc/media/base/mediacommon.h (right):

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/base/mediac...
webrtc/media/base/mediacommon.h:23: };
This is never used anywhere, and the only references to mediacommon.h are in the
files you are moving/changing.  Can we simply remove the file and all references
to it?

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/base/mutedv...
File webrtc/media/base/mutedvideocapturer.cc (right):

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/base/mutedv...
webrtc/media/base/mutedvideocapturer.cc:2: // some reason.
I think we can remove this now.

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/media.gyp
File webrtc/media/media.gyp (right):

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/media.gyp#n...
webrtc/media/media.gyp:110: 'webrtc/webrtcvoiceengine.h',
Would it be possible to make it webrtc/media/engines instead of
webrtc/media/webrtc?  Later, I think we could move those out to other places,
but webrtc/media/webrtc just looks so ugly.

On the other hand, if it's a lot of work and delays this CL much, we can do it
later.

[Taylor Brandstetter, 2016-01-27 02:39:43]

Looks great; I just noticed some minor things.

https://codereview.webrtc.org/1587193006/diff/150001/talk/app/webrtc/java/jni...
File talk/app/webrtc/java/jni/androidmediadecoder_jni.cc (right):

https://codereview.webrtc.org/1587193006/diff/150001/talk/app/webrtc/java/jni...
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc:34: #include
"talk/app/webrtc/java/jni/androidmediacodeccommon.h"  // No not sort.
What does "no not sort" mean?

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/base/crypto...
File webrtc/media/base/cryptoparams.h (right):

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/base/crypto...
webrtc/media/base/cryptoparams.h:16: namespace cricket {
On 2016/01/27 01:34:09, pthatcher1 wrote:
> Some of these files show up as adds instead of moves.  Any idea why?

In this case it's because the copyright header took up half the file, and it was
replaced with an updated one ("The WebRTC Project Authors" instead of "Google
Inc.").

Maybe we could update the copyright headers in a separate CL, so the file
history is retained?

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/media_tests...
File webrtc/media/media_tests.gypi (right):

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/media_tests...
webrtc/media/media_tests.gypi:13: 'target_name': 'rtc_unittest_main',
The name "rtc_unittest_main" makes me confuse this with the "rtc_unittests"
target. Would it make more sense to call it "rtc_media_unittest_main"?

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/media_tests...
webrtc/media/media_tests.gypi:72: 'target_name': 'libjingle_media_unittest',
Should this be rtc_media_unittests, to match the comment below?

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/webrtc_tests.gypi
File webrtc/webrtc_tests.gypi (right):

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/webrtc_tests.gypi...
webrtc/webrtc_tests.gypi:354: 'target_name': 'libjingle_media_unittest_run',
Should this also be "rtc" instead of "libjingle"?

[pthatcher, 2016-01-27 21:50:57]

pthatcher@google.com changed reviewers:
+ pthatcher@google.com

[pthatcher, 2016-01-27 21:50:59]

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/base/crypto...
File webrtc/media/base/cryptoparams.h (right):

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/base/crypto...
webrtc/media/base/cryptoparams.h:16: namespace cricket {
On 2016/01/27 02:39:42, Taylor Brandstetter wrote:
> On 2016/01/27 01:34:09, pthatcher1 wrote:
> > Some of these files show up as adds instead of moves.  Any idea why?
> 
> In this case it's because the copyright header took up half the file, and it
was
> replaced with an updated one ("The WebRTC Project Authors" instead of "Google
> Inc.").
> 
> Maybe we could update the copyright headers in a separate CL, so the file
> history is retained?

If that's what it takes to keep the history intact, let's do it.

[kjellander_webrtc, 2016-02-04 10:18:24]

Patchset #10 (id:160001) has been deleted

[kjellander_webrtc, 2016-02-04 10:18:54]

Patchset #10 (id:160001) has been deleted

[kjellander_webrtc, 2016-02-04 21:29:13]

Description was changed from

==========
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/.

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
==========

to

==========
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
==========

[kjellander_webrtc, 2016-02-04 21:50:47]

The CQ bit was checked by kjellander@webrtc.org to run a CQ dry run

[commit-bot: I haz the power, 2016-02-04 21:51:03]

Dry run: CQ is trying da patch. Follow status at
 https://chromium-cq-status.appspot.com/patch-status/1587193006/180001
View timeline at
 https://chromium-cq-status.appspot.com/patch-timeline/1587193006/180001

[kjellander_webrtc, 2016-02-04 21:52:10]

OK I'm finally ready for another round after spending quite some time rebasing
this properly. I'm writing this before seeing all the trybots pass, but if some
are red it's probably a minor thing to fix.

Please have a look so we can get this in asap!

https://codereview.webrtc.org/1587193006/diff/150001/talk/app/webrtc/java/jni...
File talk/app/webrtc/java/jni/androidmediadecoder_jni.cc (right):

https://codereview.webrtc.org/1587193006/diff/150001/talk/app/webrtc/java/jni...
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc:34: #include
"talk/app/webrtc/java/jni/androidmediacodeccommon.h"  // No not sort.
On 2016/01/27 02:39:42, Taylor Brandstetter wrote:
> What does "no not sort" mean?

Ah, bad typo there. I ran into build errors when sorting these headers
alphabetically. Tte comment was supposed to be "Do not sort.", but I changed it
to a more descriptive comment now.

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/base/crypto...
File webrtc/media/base/cryptoparams.h (right):

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/base/crypto...
webrtc/media/base/cryptoparams.h:16: namespace cricket {
On 2016/01/27 21:50:59, pthatcher wrote:
> On 2016/01/27 02:39:42, Taylor Brandstetter wrote:
> > On 2016/01/27 01:34:09, pthatcher1 wrote:
> > > Some of these files show up as adds instead of moves.  Any idea why?
> > 
> > In this case it's because the copyright header took up half the file, and it
> was
> > replaced with an updated one ("The WebRTC Project Authors" instead of
"Google
> > Inc.").
> > 
> > Maybe we could update the copyright headers in a separate CL, so the file
> > history is retained?
> 
> If that's what it takes to keep the history intact, let's do it.

Good point, I changed the back again (since I wrote a script to do these changes
it was pretty easy).
I'll do a follow-up CL after landing this, with the license change.

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/base/mediac...
File webrtc/media/base/mediacommon.h (right):

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/base/mediac...
webrtc/media/base/mediacommon.h:23: };
On 2016/01/27 01:34:09, pthatcher1 wrote:
> This is never used anywhere, and the only references to mediacommon.h are in
the
> files you are moving/changing.  Can we simply remove the file and all
references
> to it?

Sure, but I still would like to do that in a separate CL. I'll add it to my TODO
for this effort.

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/base/mutedv...
File webrtc/media/base/mutedvideocapturer.cc (right):

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/base/mutedv...
webrtc/media/base/mutedvideocapturer.cc:2: // some reason.
On 2016/01/27 01:34:09, pthatcher1 wrote:
> I think we can remove this now.

I'll do it in a separate CL though, made a note.

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/media.gyp
File webrtc/media/media.gyp (right):

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/media.gyp#n...
webrtc/media/media.gyp:110: 'webrtc/webrtcvoiceengine.h',
On 2016/01/27 01:34:09, pthatcher1 wrote:
> Would it be possible to make it webrtc/media/engines instead of
> webrtc/media/webrtc?  Later, I think we could move those out to other places,
> but webrtc/media/webrtc just looks so ugly.
> 
> On the other hand, if it's a lot of work and delays this CL much, we can do it
> later.

Yeah, it'll be even more renaming and updating include paths. I can do it in a
later CL. Made a note about renaming webrtc/media/webrtc -> webrtc/media/engines

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/media_tests...
File webrtc/media/media_tests.gypi (right):

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/media_tests...
webrtc/media/media_tests.gypi:13: 'target_name': 'rtc_unittest_main',
On 2016/01/27 02:39:43, Taylor Brandstetter wrote:
> The name "rtc_unittest_main" makes me confuse this with the "rtc_unittests"
> target. Would it make more sense to call it "rtc_media_unittest_main"?

I'm not sure what to name it, it's depended upon by peerconnection_unittests and
libjingle_p2p_unittest as well so that won't be an improvement. Let's keep it as
is for this CL.

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/media/media_tests...
webrtc/media/media_tests.gypi:72: 'target_name': 'libjingle_media_unittest',
On 2016/01/27 02:39:42, Taylor Brandstetter wrote:
> Should this be rtc_media_unittests, to match the comment below?

Yes, but in order to have passing trybots for this CL I need to keep it
unchanged. I want to be sure not to break the test. Then I can rename the test +
update the bot configs in a separate (much smalller CL).

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/webrtc_tests.gypi
File webrtc/webrtc_tests.gypi (right):

https://codereview.webrtc.org/1587193006/diff/150001/webrtc/webrtc_tests.gypi...
webrtc/webrtc_tests.gypi:354: 'target_name': 'libjingle_media_unittest_run',
On 2016/01/27 02:39:42, Taylor Brandstetter wrote:
> Should this also be "rtc" instead of "libjingle"?

Yeah, but I'll rename the test in a later CL, so we can get trybot coverage for
this CL (since the changes potentially can break the tests).

[commit-bot: I haz the power, 2016-02-04 22:08:13]

The CQ bit was unchecked by commit-bot@chromium.org

[commit-bot: I haz the power, 2016-02-04 22:08:15]

Dry run: Try jobs failed on following builders:
  presubmit on tryserver.webrtc (JOB_FAILED,
http://build.chromium.org/p/tryserver.webrtc/builders/presubmit/builds/3250)

[Taylor Brandstetter, 2016-02-04 23:04:18]

lgtm!

[pthatcher1, 2016-02-04 23:37:09]

lgtm

[kjellander_webrtc, 2016-02-05 07:29:29]

Description was changed from

==========
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
==========

to

==========
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
TBR=tommi@webrtc.org
==========

[kjellander_webrtc, 2016-02-05 07:31:46]

On 2016/02/04 23:37:09, pthatcher1 wrote:
> lgtm

Wow, all trybots went green!
I'm TBR'ing tommi for root owners rubberstamp.
I'll coordinate will grunell@ for rolling this in to Chromium using the changes
in https://codereview.chromium.org/1604303002/

[kjellander_webrtc, 2016-02-05 07:32:04]

The CQ bit was checked by kjellander@webrtc.org

[commit-bot: I haz the power, 2016-02-05 07:32:18]

CQ is trying da patch. Follow status at
 https://chromium-cq-status.appspot.com/patch-status/1587193006/180001
View timeline at
 https://chromium-cq-status.appspot.com/patch-timeline/1587193006/180001

[kjellander_webrtc, 2016-02-05 07:37:40]

Description was changed from

==========
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
TBR=tommi@webrtc.org
==========

to

==========
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
==========

[kjellander_webrtc, 2016-02-05 07:38:51]

On 2016/02/05 07:32:18, commit-bot: I haz the power wrote:
> CQ is trying da patch. Follow status at
>  https://chromium-cq-status.appspot.com/patch-status/1587193006/180001
> View timeline at
>  https://chromium-cq-status.appspot.com/patch-timeline/1587193006/180001

Bypassing presubmit with NOPRESUBMIT=True since there are many cpplint errors
and the licenses that fails the presubmit check.

[commit-bot: I haz the power, 2016-02-05 07:41:34]

The CQ bit was unchecked by commit-bot@chromium.org

[commit-bot: I haz the power, 2016-02-05 07:41:36]

Try jobs failed on following builders:
  presubmit on tryserver.webrtc (JOB_FAILED,
http://build.chromium.org/p/tryserver.webrtc/builders/presubmit/builds/3255)

[kjellander_webrtc, 2016-02-05 07:51:10]

The CQ bit was checked by kjellander@webrtc.org

[commit-bot: I haz the power, 2016-02-05 07:51:23]

CQ is trying da patch. Follow status at
 https://chromium-cq-status.appspot.com/patch-status/1587193006/180001
View timeline at
 https://chromium-cq-status.appspot.com/patch-timeline/1587193006/180001

[commit-bot: I haz the power, 2016-02-05 07:52:34]

Description was changed from

==========
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
==========

to

==========
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
==========

[commit-bot: I haz the power, 2016-02-05 07:52:36]

Committed patchset #11 (id:180001)

[commit-bot: I haz the power, 2016-02-05 07:52:40]

Description was changed from

==========
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
==========

to

==========
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

Committed: https://crrev.com/a96e2d77cb80174af76fa3e60d7c98f6ab565f57
Cr-Commit-Position: refs/heads/master@{#11495}
==========

[commit-bot: I haz the power, 2016-02-05 07:52:41]

Patchset 11 (id:??) landed as
https://crrev.com/a96e2d77cb80174af76fa3e60d7c98f6ab565f57
Cr-Commit-Position: refs/heads/master@{#11495}

[tommi, 2016-02-05 08:16:34]

lgtm - hope it sticks! :)