20baeb4811
Bug 1404039 - Add a unittest for VideoConduit getting a signal to reduce quality due to load or bandwidth; r=pehrsons
...
MozReview-Commit-ID: 5J3wINSnStR
--HG--
extra : rebase_source : 0c9518c0501ca579ed9d948bde63159c21d9657c
2017-10-18 15:19:21 -04:00
858f178676
Bug 1410090 - Remove remnant of FFOS OMX code. r=cpearce
...
For webrtc, the most important part of the code had already been removed in bug 1355048 and could no longer be called
MozReview-Commit-ID: Fx9XI0zR1gn
--HG--
extra : rebase_source : 360996760abab650684440fbeea258b43dccfd83
2017-10-19 16:51:04 +02:00
a098eb0deb
Bug 1408504 - Implement WriteSamples in tests for big endian platforms; r=dminor
...
Webrtc code supports big endian platforms, but a method in tests lacks a big endian
variant. Add it based on WavReader/WavWriter code.
MozReview-Commit-ID: A4OTnYlGgvU
--HG--
extra : rebase_source : f331c799cea89e6090fd02269d3ee8728cbeca45
2017-10-17 10:27:12 +00:00
5ea6a54fa2
Bug 1402495 - changes to support MID in audio packets. r=drno
...
MozReview-Commit-ID: BivuIsgNLYI
--HG--
extra : rebase_source : a5b08a2dd4c8c19985bd85f77679e0300c15589f
2017-09-15 13:55:17 -05:00
cd290b865f
Bug 1402495 - changes to support MID in video packets. r=drno
...
MozReview-Commit-ID: E7RoFZBb5C8
--HG--
extra : rebase_source : 863b94fe968f55f86c19b13073b831326e5387bb
2017-08-18 09:53:34 -05:00
eb9a2ed0f2
Merge inbound to central, a=merge
...
MozReview-Commit-ID: IqwKWn7ceHC
2017-09-29 14:47:25 -07:00
5af93e95a8
Bug 1402818 - Set hwnd_ to NULL in PlatformUIThread::Stop(); r=pehrsons
...
We only create an event window in Start() if hwnd_ is NULL, but Stop() does
not set it to NULL. This causes the thread to no longer be a GUI thread if
Stop() and then Start() is called on the same thread leading to assertion
failures.
MozReview-Commit-ID: 5TpazUCqBuR
--HG--
extra : rebase_source : 461066e576af87b27c82075a15d6b5772282b54c
2017-09-28 16:32:31 -04:00
4658732fb4
Bug 1382433 - use fixed fps instead of floating fps to avoid very low fps. r=jib
...
MozReview-Commit-ID: 2QUWJM2LNkK
--HG--
extra : rebase_source : 5e323539c0d600979726d552938c9f06ed5cf5e2
2017-09-27 18:50:54 +08:00
b73bb761aa
Bug 1402348 - fix webrtc.org screen_capturer_unittest and window_capturer_unittest segfault on Linux; r=jesup
...
We weren't defining USE_X11 when building the tests resulting in inconsistent
versions of the headers depending upon where they were compiled.
MozReview-Commit-ID: 298yRvIjXgb
--HG--
extra : rebase_source : 7406d14db3b41b8b5f579cc572ccb283064dbe29
2017-09-22 14:33:49 -04:00
30d76b125c
Bug 1382182 - Build jsep using moz.build; r=ted,jesup
...
MozReview-Commit-ID: 9UhlBZi0yO0
--HG--
extra : rebase_source : 7ad69af2dfb61ad88fc5f87f4d17a09abbf23edc
2017-07-20 11:24:08 -04:00
800b45250d
Bug 1402242: use 2-byte vp8 picture ids always (cherrypick of issue 7877) r=dminor
2017-09-22 14:16:59 -04:00
979c34dc04
Bug 1389534 - Enable continuous auto focus mode. r=jib
...
MozReview-Commit-ID: DrTkb9CxjlA
--HG--
extra : rebase_source : c65aabcf0939a20814690a4c3349846fde0be00a
2017-09-07 18:25:45 +08:00
a531d080c3
Bug 1393095 - remote audio receiver stats missing;r=dminor,jesup
...
MozReview-Commit-ID: 9izPPOqybcK
--HG--
extra : rebase_source : 31578d7198929267e423a1c0c4b13cd49f110629
2017-09-13 01:38:35 -07:00
c9a33a6e73
Bug 1393687 - Fix handling of max-fr parameter; r=jesup
...
MozReview-Commit-ID: C30K1Pogm5u
--HG--
extra : rebase_source : e53ed34aed0c59b1495581562901b4e7cc78e3b0
2017-09-07 16:05:52 -04:00
c0a229d4c3
Bug 1386876 - Replace all uses of DISABLE_STL_WRAPPING with a template, remove DISABLE_STL_WRAPPING. r=glandium
...
MozReview-Commit-ID: FMEtb5PY7iP
--HG--
extra : rebase_source : 3cdee7528846462c758e623d6bcd2e6e17dbabff
2017-09-11 11:33:26 -07:00
28d9a3c000
Bug 1395566 - Enable more webrtc.org desktop capture tests; r=jesup
...
This adds the remaining desktop_capture unit tests with a few exceptions:
app_capturer_unittest does not compile and is not built by webrtc.org,
and desktop_capturer_differ_wrapper_unittest, rgba_color_unittest,
screen_drawer_unittest and test_utils_unittest rely upon code that we do
not build.
--HG--
extra : rebase_source : 6bdac36a46723ade37d6c2ba4a9384ff7205a6e1
2017-09-01 09:44:29 -04:00
f1506a2ce3
Bug 1395849 - Fix G.722 audio codec; r=jesup
...
Even though we were building the G.722 codec, we weren't setting the
defines so that it would actually be used.
MozReview-Commit-ID: Dw8l2sYwZFA
--HG--
extra : rebase_source : 10cbd61cb483536be32d7b40f1e64222c3259089
2017-09-01 14:48:36 -04:00
8b84853ed6
Merge m-c to autoland, a=merge
...
MozReview-Commit-ID: GcHZLNPPNnI
2017-09-01 16:34:14 -07:00
ed140d9eeb
Bug 1395289 - Convert from points to pixels when invalidating rect for screensharing when building on OS X older than 10.8; r=jesup
...
When we build for versions of OS X below 10.8 it is still necessary to
convert from points to pixels when invalidating rectangles as we use the
CGRegisterScreenRefreshCallback and CGRegisterMoveCallback interfaces.
--HG--
extra : rebase_source : cc3a405c1faaf010922b9acbe0edc21da15bb9ac
2017-08-31 13:29:34 -04:00
a03ea28fe8
Bug 1395769 - Don't define MOZ_JEMALLOC_IMPL when building webrtc-gtest. r=froydnj
...
This define is only meant to be set when building mozjemalloc itself.
--HG--
extra : rebase_source : 5660b691855c3b0be55375ad8d9525ea2288bb69
2017-08-31 16:16:23 +09:00
c756f0c92e
Bug 1393795 Lowercase includes so WebRTC compiles with MinGW r=jesup
...
This edits the third party SCTP library, but upstream has already applied this change to master
MozReview-Commit-ID: ERpMc8EvYZ7
--HG--
extra : rebase_source : cea0d3758275b73a395ad2738edd8eb57c833e1a
2017-08-25 14:04:36 -05:00
f7fa48a68d
Bug 1392515 - Properly link webrtc-gtest on Linux. r=gps
...
The main difference between Linux and other platforms is that mozglue is
a static library and doesn't include the allocator, while it is a
dynamic library and includes the allocator on other platforms.
As such, linking against mozglue alone doesn't guarantee everything that
should be linked is linked, which GeckoProgram() does. But since
webrtc-gtest doesn't want to link libxul, we use linkage=None.
--HG--
extra : rebase_source : 552b123cb4ef6a861a49bc5eea0f03b9b5427e8b
2017-08-22 14:51:44 +09:00
6dd42e2664
Merge inbound to central, a=merge
...
MozReview-Commit-ID: BMWuqvmTljV
2017-08-22 17:07:23 -07:00
b779bb7db1
Bug 1392583 - Remove calls to FATAL() from audio_device_template.h for unimplemented functions; r=padenot
...
This removes calls to FATAL() and replaces them with LOG() statements. The
unimplemented functions already returned error codes.
MozReview-Commit-ID: KgXVCIKWoLp
--HG--
extra : rebase_source : 993d3700e734fd6042fc5708261dc58d9987e64b
2017-08-22 09:58:03 -04:00
0c9287d1b6
Bug 1388129 - Fix interaction between quality scaler and scaleResolutionDownBy; r=jesup
...
Currently we apply the scaleResolutionDownBy factor to the resolution
requested by the quality scaler. This can lead to a cycle where the quality
scaler requests a slightly larger resolution and we scale it down even
smaller than the current resolution.
This changes things so that we only apply scaleResolutionDownBy to the
incoming resolution and then take the minimum of the scaled resolution and
the resolution requested by the quality scaler.
--HG--
extra : rebase_source : 1d16ed60b575c48d43e2e1928c518bc197339cbc
2017-08-18 12:11:50 -04:00
b1fc5e008c
Merge inbound to central, a=merge
...
MozReview-Commit-ID: 4cWGBbMEU2x
2017-08-18 15:53:07 -07:00
50513900c7
Bug 1389598 - Part 2: Remove gonk references from media/ r=jesup
...
--HG--
extra : rebase_source : d1af2d0987038e1c0b0b0c971d0d2e4e9f08364a
2017-08-11 17:46:15 -07:00
cb5753f469
Bug 1390318 - add MID support to webrtc.org. r=drno
...
MozReview-Commit-ID: EHgEuhw855n
--HG--
extra : rebase_source : 8d575753a628b18472c3acd13ca88f5aa63c16b1
2017-07-27 16:19:56 -05:00
9f34033ad7
Bug 1387525 - Fix crash in webrtc::NetEqImpl::InsertPacketInternal; r=jesup
...
The webrtc.org code assumes we will always get a valid decoder for a known payload
type, but this is not true for our builds. This adds a check that we have a valid
decoder before calling IncomingPacket.
MozReview-Commit-ID: GUJR7Qn28vh
--HG--
extra : rebase_source : 6bd5872b59d964c3246708f0e6f549bb74dcc0b3
2017-08-15 08:25:43 -04:00
830c799f81
Bug 1389256 - fix incomplete handling of RepairedRtpStreamId after webrtc.org backport of RtpStreamId. r=bwc
...
During my backport of RtpStreamId from webrtc.org, I missed a few
places where RepairedRtpStreamId was used or was not completely
implemented. Also, the webrtc.org code used repairedStreamId,
which is not really correct per the spec (draft-ietf-avtext-rid)
so I fixed all occurances to use the correct repairedRtpStreamId
to avoid confusion later.
The RTP header extensions default IDs for RtpStreamId and
RepairedRtpStreamId were also adjusted to not collide with
PlayoutDelay's default ID.
MozReview-Commit-ID: HSlS8nsKQ29
--HG--
extra : rebase_source : f1bf7fc9ceec22de1c56ef3b7be22fccea01ecdb
2017-07-28 14:52:46 -05:00
09ca1b1297
Bug 1384655 - Remove obsolete latency measuring macros in downstream code we don't use anymore. r=jesup
...
MozReview-Commit-ID: ARAWGe7yFM4
--HG--
extra : amend_source : b1b3f4f33c9c2b15465ac5670a3639b8527a16a3
2017-08-08 10:00:37 +02:00
d2130a3810
Bug 1386164 - Part 2. Use MOZ_SYSTEM_LIBEVENT for webrtc. r=jesup
...
GYP of WebRTC should reference MOZ_SYSTEM_LIBEVENT values if available.
MozReview-Commit-ID: CshsPrRidM8
--HG--
extra : rebase_source : 9e619c2f49e7c2b3f680814b95b823996773fa6c
2017-08-03 13:28:25 +09:00
b2458fa168
Bug 1386164 - Part 1. Use libevent2 headers instead of deprecated event.h. r=jesup
...
libevent uses event.h header that is fuzzy name. Since our in-tree libevent is libevent2, we should use libevent2 headers instead of deprecated event.h
MozReview-Commit-ID: 6DjW9JEkNWL
--HG--
extra : rebase_source : b774e177b137bf7427122253a3e4c698689e08a4
2017-08-03 13:22:26 +09:00
c6d57aa7de
Bug 1384874 - Fix build failure on OSX 10.11.6 after Bug 1368030 landed; r=mjf
...
--HG--
extra : rebase_source : 36569545916cd2a201908ca835efc601ffb49f4d
2017-07-27 09:38:49 -04:00
460bb5228f
Bug 1382681 - Remove java.lang.RuntimeException in VideoCaptureAndroid; r=jesup
...
There does not seem to be any bad effects from calling stop twice, so just log that
it has happened rather than throwing an exception.
--HG--
extra : rebase_source : 0d92bad7b33010f50f41de8498b8406c3521c9e7
2017-07-25 11:31:26 -04:00
fa12aab7f8
Bug 1384826 - Media: WebRTC: Fix build config for MIPS. r=jesup
2017-07-26 23:12:00 -04:00
d360d49d2a
merge mozilla-inbound to mozilla-central a=merge
2017-07-27 10:57:25 +02:00
01192ea8bd
Bug 1368030 - Fix race condition in ScreenCapturerMac. r=jesup
...
The race condition is between ~ScreenCapturerMac and the ScreenRefresh and
ScreenUpdateMove callbacks. The destructor calls
UnregisterRefreshAndMoveHandlers but a callback may still occur after the
destruction of the object.
Rather than passing a pointer to ScreenCapturerMac into the callback, this
adds a separate object which keeps a pointer to ScreenCapturerMac guarded
by a CriticalSection. The destructor sets the ScreenCapturerMac to nullptr.
In the next callback, the handler unregisters the callbacks and deletes
the object.
The downside to this approach is that if the ScreenCapturerMac
object is allocated and deallocated before a callback occurs, the memory
for the separate object will be leaked.
2017-07-19 14:49:05 -04:00
ba4b3b9101
Bug 1384233 - Remove SizePrintfMacros.h. r=froydnj
...
We have a minimum requirement of VS 2015 for Windows builds, which supports
the z length modifier for format specifiers. So we don't need SizePrintfMacros.h
any more, and can just use %zu and friends directly everywhere.
MozReview-Commit-ID: 6s78RvPFMzv
--HG--
extra : rebase_source : 009ea39eb4dac1c927aa03e4f97d8ab673de8a0e
2017-07-26 16:03:57 -04:00
ffa6f2f3c0
Bug 1377959 - fix compiler warning about varargs functions; r=jesup
2017-07-25 16:52:56 -04:00
115784405b
Backed out changeset 0a60cc198321 for browser_devices_get_user_media_screen.js | application crashed [@ mozalloc_abort(char const*)]
2017-07-25 17:40:59 +02:00
6e3ccd3e2a
Bug 1368030 - Fix race condition in ScreenCapturerMac. r=jesup
...
The race condition is between ~ScreenCapturerMac and the ScreenRefresh and
ScreenUpdateMove callbacks. The destructor calls
UnregisterRefreshAndMoveHandlers but a callback may still occur after the
destruction of the object.
Rather than passing a pointer to ScreenCapturerMac into the callback, this
adds a separate object which keeps a pointer to ScreenCapturerMac guarded
by a CriticalSection. The destructor sets the ScreenCapturerMac to nullptr.
In the next callback, the handler unregisters the callbacks and deletes
the object.
The downside to this approach is that if the ScreenCapturerMac
object is allocated and deallocated before a callback occurs, the memory
for the separate object will be leaked.
2017-07-19 14:49:05 -04:00
ac3e6bddb4
Bug 1382099 - Remove MOZ_WIDGET_GONK from media/, uriloader/, widget, /xpfe/. r=snorp.
...
--HG--
extra : rebase_source : 75fe5b8320d52c7316ca547f706b64f30250d28c
2017-07-24 10:08:55 +10:00
ab82f57025
Bug 1383272 - fixing incorrect comparison in RtpStreamId::Parse(...) r=bwc
...
When adding the length check for parsing RtpStreamId, I incorrectly used
the '<=' operator instead of the '>' operator.
MozReview-Commit-ID: 46XZBqWxkBc
--HG--
extra : rebase_source : 6290aeed489770070308aafacad01ce5b63a60a1
2017-07-21 15:54:11 -05:00
5f43d20a48
Bug 1379392 - Avoid double-delete on failure to init VideoCapture module. r=mchiang
...
MozReview-Commit-ID: I9p2NVzqc8
2017-07-19 16:14:47 -04:00
cf76a1f916
Bug 1380430 - Backport current webrtc.org RtpHeaderExtension handling changes and RtpStreamId implementation. r=drno
...
The new RtpHeaderExtension handling works better with variable length
header extensions, and the new RtpStreamId implementation takes
advantage of it. This is useful to us because we'll be able to add
MID support using the same mechanism.
MozReview-Commit-ID: 5VYQYvhD5gr
--HG--
extra : rebase_source : 900126e0b136343a2767715b12d906b1dbbabe36
2017-07-12 13:44:40 -05:00
817491807e
Bug 1379836 - Fix AEC Logging; r=jesup
...
This enables apm logging by setting the apm_debug_dump variable in gyp.mozbuild.
Prior to this change, some files were including apm_data_dumper.h with logging
enabled and some were not.
This also removes the AEC* C interface and calls into webrtc::Trace directly.
Whatever historical reasons for having a C interface into these calls no
longer seems to apply. In addition reserving a buffer for the base file name
and then ensuring it was null terminated caused an ASAN "stack-buffer-overflow"
while testing. This was because it was not handling an empty base file
name properly. This would not normally happen if AEC logging was enabled through
about:webrtc, but it still seems safer to use std::string.
MozReview-Commit-ID: Ikz5xO74syA
--HG--
extra : rebase_source : 8e0c59117135fadb75f4a7e6be5588af1404533d
2017-07-12 16:49:15 -04:00
112117877f
Bug 1379743 - Recalculate stride when scaling desktop capture; r=jesup
...
Currently we calculate the stride prior to calculating the scaled dimensions
which results in garbage video when scaling the input frame. This recalculates
the stride based upon the scaled dimensions.
MozReview-Commit-ID: BwOlFwzqdco
--HG--
extra : rebase_source : df9aab6dea81055ca557ba9ea0a9700f7347f389
extra : amend_source : 79e14700aeb5975f6303bc021d62c7f322d298db
2017-07-11 11:55:19 -04:00
d2e8774b67
Bug 1378412 - Fix build error on Linux 32 bit due to a warning in task_queue_libevent.cc with clang 4.0; r=jesup
...
MozReview-Commit-ID: 5GW5CJMV7V5
--HG--
extra : rebase_source : df946b6a343a9b2b57224a62ad01c43a21337a33
2017-07-06 09:48:28 -04:00
22cb9f77bb
Merge m-c to m-i
...
MozReview-Commit-ID: H6zGgEm7oOM
2017-07-04 20:32:07 -07:00
b003a6a704
Bug 1377803 - Remove an unnecessary plarena.h #include in WebRTC. r=drno,glandium.
...
We can avoid the symbol visibility problem by putting
sanitizer/asan_interface.h in the config/system-headers.
--HG--
extra : rebase_source : bc81a81ef8970c3544febf06631740208583c7fa
2017-07-04 09:58:38 +10:00
9f97858792
Bug 1375238: cast so 32-bit clang will compile r=dminor
2017-06-29 14:24:41 -07:00
a1ecc018be
Bug 1377078 - removed '//' from #include path, r=jesup
...
MozReview-Commit-ID: 3m0tAkYRpom
--HG--
extra : rebase_source : 014bb4455e8454f23de6ecd903199ccddcc809b0
2017-06-29 11:15:14 +02:00
5579381f1c
Bug 1305813 - do not send empty StreamId;r=drno
...
Also moving RID (StreamID) storage to the stack to clear up the TODO item
and simplify the code.
When unset the StreamID is stored as an empty string.
Added missing GUARDED_BY on rid_
Added a check to shortcut checking strlen on the StreamId when it is an empty string (which is most of the time).
MozReview-Commit-ID: EPUlPNBXYsQ
--HG--
extra : rebase_source : 08e1b9ea796c991d141164424014d2311ff9341c
2017-06-22 00:46:07 -07:00
14928e4c02
Bug 1374938 - use our own avfoundation wrapper. r=jib
...
MozReview-Commit-ID: KObTvtvRk10
--HG--
extra : rebase_source : 0fe3d7580cd6f64fad2dce791d52965893a7046a
2017-06-22 14:22:10 +08:00
1244053765
Bug 1373525 - webrtc: Remove unused member function in WrappableJSErrorResult. r=bwc
...
media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp:181:3 [-Wunused-member-function] unused member function 'operator (anonymous namespace)::JSErrorResult &'
And suppress -Wcomma warnings in upstream webrtc code:
media/webrtc/trunk/webrtc/modules/audio_coding/neteq/background_noise.h:98:22: warning: possible misuse of comma operator here [-Wcomma]
media/webrtc/trunk/webrtc/modules/desktop_capture/differ_unittest.cc:187:22: warning: possible misuse of comma operator here [-Wcomma]
MozReview-Commit-ID: FVecnczsWk7
--HG--
extra : source : a651d94c9adcd64690a6acba4629cf7e1b299e3c
extra : intermediate-source : d5cdb25590475e306cdb8b9766a237e22940f7fa
2017-06-12 21:43:29 -07:00
76c049c746
Bug 1373988 - Make webrtc-gtest build work with system jpeg and libvpx; r=glandium
...
MozReview-Commit-ID: 1x8v7G2fvlK
--HG--
extra : rebase_source : cf5475a104b95901cc66045e7f8a61d82f397980
2017-06-19 08:56:19 -04:00
897f2d691d
Bug 1341285 - Restore number of CPU cores detection on BSDs. r=jesup
2017-06-17 13:43:31 -04:00
7bf7ce4954
Bug 1341285 - Sync sndio with WebRTC 57 and fix warnings. r=jesup
2017-06-17 13:43:11 -04:00
1b26da1b2f
merge mozilla-central to mozilla-inbound. r=merge a=merge
2017-06-15 11:17:07 +02:00
8e114807a7
Bug 1341285 - Fix lint errors in WebRtcAudioTrack.java; r=drno
...
--HG--
extra : rebase_source : 84981127cde93ca9de53d56ee29a4d8b00aebfeb
2017-06-14 08:38:07 -04:00
ccc649dbc7
Bug 1341285 - Add missing BSD bits lost during the rebase. r=jesup
2017-06-14 20:58:52 -04:00
6b92cb978e
Bug 1341285: Fix bustage on android due to merge failure r=bustage
2017-06-13 02:33:20 -04:00
450c4d90a1
Bug 1341285: rollup of changes for webrtc after applying webrtc.org v57 update r=ng,jesup,pehrsons,drno,dminor,cpearce,jya,glandium,dmajor
...
Includes re-importing gyp files removed from upstream in v56, and then
updating them to match the BUILD.gn file changes.
--HG--
rename : media/webrtc/trunk/webrtc/call/audio_send_stream.cc => media/webrtc/trunk/webrtc/call/audio_send_stream_call.cc
2017-06-13 01:54:13 -04:00
c2dfe0a8c9
Bug 1341285: Webrtc updated to branch 57 rev 52b6562a10b495; initial pull made Feb 2 2017 14:38 EST
...
Pull updated from 71394677e4dc343ca5c0f996037207a9bd9616c9 to 52b6562a10b495 in late May
--HG--
rename : media/webrtc/trunk/webrtc/base/iosfilesystem.mm => media/webrtc/trunk/webrtc/base/applefilesystem.mm
rename : media/webrtc/trunk/webrtc/test/testsupport/gtest_prod_util.h => media/webrtc/trunk/webrtc/base/gtest_prod_util.h
rename : media/webrtc/trunk/webrtc/base/exp_filter.cc => media/webrtc/trunk/webrtc/base/numerics/exp_filter.cc
rename : media/webrtc/trunk/webrtc/base/exp_filter.h => media/webrtc/trunk/webrtc/base/numerics/exp_filter.h
rename : media/webrtc/trunk/webrtc/base/exp_filter_unittest.cc => media/webrtc/trunk/webrtc/base/numerics/exp_filter_unittest.cc
rename : media/webrtc/trunk/webrtc/base/rtccertificate_unittests.cc => media/webrtc/trunk/webrtc/base/rtccertificate_unittest.cc
rename : media/webrtc/trunk/webrtc/common_audio/swap_queue.h => media/webrtc/trunk/webrtc/base/swap_queue.h
rename : media/webrtc/trunk/webrtc/common_audio/swap_queue_unittest.cc => media/webrtc/trunk/webrtc/base/swap_queue_unittest.cc
rename : media/webrtc/trunk/webrtc/audio_receive_stream.h => media/webrtc/trunk/webrtc/call/audio_receive_stream.h
rename : media/webrtc/trunk/webrtc/audio_state.h => media/webrtc/trunk/webrtc/call/audio_state.h
rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser_unittest.cc => media/webrtc/trunk/webrtc/common_video/h264/h264_bitstream_parser_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_sps_parser_unittest.cc => media/webrtc/trunk/webrtc/common_video/h264/sps_parser_unittest.cc
rename : media/webrtc/trunk/webrtc/frame_callback.h => media/webrtc/trunk/webrtc/common_video/include/frame_callback.h
rename : media/webrtc/trunk/webrtc/call/rtc_event_log.proto => media/webrtc/trunk/webrtc/logging/rtc_event_log/rtc_event_log.proto
rename : media/webrtc/trunk/webrtc/video/video_decoder.cc => media/webrtc/trunk/webrtc/media/engine/videodecodersoftwarefallbackwrapper.cc
rename : media/webrtc/trunk/webrtc/video/video_encoder_unittest.cc => media/webrtc/trunk/webrtc/media/engine/videoencodersoftwarefallbackwrapper_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack_tracker.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack_tracker.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack_tracker_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_neon.c => media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_neon.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_resampler.c => media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_resampler.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation.c => media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core.c => media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_c.c => media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_c.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_mips.c => media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_mips.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_neon.c => media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_neon.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aecm/echo_control_mobile.c => media/webrtc/trunk/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/histogram.h => media/webrtc/trunk/webrtc/modules/audio_processing/agc/loudness_histogram.h
rename : media/webrtc/trunk/webrtc/modules/audio_processing/test/audio_processing_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_unittest.cc
rename : media/webrtc/trunk/webrtc/test/common_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_processing/config_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/utility/delay_estimator.c => media/webrtc/trunk/webrtc/modules/audio_processing/utility/delay_estimator.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.c => media/webrtc/trunk/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_rdft_mips.c => media/webrtc/trunk/webrtc/modules/audio_processing/utility/ooura_fft_mips.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_rdft_neon.c => media/webrtc/trunk/webrtc/modules/audio_processing/utility/ooura_fft_neon.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_rdft_sse2.c => media/webrtc/trunk/webrtc/modules/audio_processing/utility/ooura_fft_sse2.cc
rename : media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios.h => media/webrtc/trunk/webrtc/modules/video_capture/objc/device_info.h
rename : media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios.mm => media/webrtc/trunk/webrtc/modules/video_capture/objc/device_info.mm
rename : media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios_objc.h => media/webrtc/trunk/webrtc/modules/video_capture/objc/device_info_objc.h
rename : media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios_objc.mm => media/webrtc/trunk/webrtc/modules/video_capture/objc/device_info_objc.mm
rename : media/webrtc/trunk/webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.h => media/webrtc/trunk/webrtc/modules/video_capture/objc/rtc_video_capture_objc.h
rename : media/webrtc/trunk/webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.mm => media/webrtc/trunk/webrtc/modules/video_capture/objc/rtc_video_capture_objc.mm
rename : media/webrtc/trunk/webrtc/modules/video_capture/ios/video_capture_ios.h => media/webrtc/trunk/webrtc/modules/video_capture/objc/video_capture.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/screenshare_layers_unittest.cc => media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_screenshare_layers_unittest.cc
rename : media/webrtc/trunk/webrtc/p2p/base/constants.cc => media/webrtc/trunk/webrtc/p2p/base/p2pconstants.cc
rename : media/webrtc/trunk/webrtc/p2p/base/constants.h => media/webrtc/trunk/webrtc/p2p/base/p2pconstants.h
rename : media/webrtc/trunk/webrtc/base/objc/NSString+StdString.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/NSString+StdString.h
rename : media/webrtc/trunk/webrtc/base/objc/NSString+StdString.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/NSString+StdString.mm
rename : media/webrtc/trunk/webrtc/base/objc/RTCCameraPreviewView.m => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCCameraPreviewView.m
rename : media/webrtc/trunk/webrtc/base/objc/RTCDispatcher.m => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCDispatcher.m
rename : media/webrtc/trunk/webrtc/api/objc/RTCEAGLVideoView.m => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCEAGLVideoView.m
rename : media/webrtc/trunk/webrtc/api/objc/RTCIceCandidate+Private.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCIceCandidate+Private.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCIceCandidate.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCIceCandidate.mm
rename : media/webrtc/trunk/webrtc/api/objc/RTCIceServer+Private.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCIceServer+Private.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCIceServer.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCIceServer.mm
rename : media/webrtc/trunk/webrtc/api/objc/RTCStatsReport+Private.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCLegacyStatsReport+Private.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCStatsReport.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCLegacyStatsReport.mm
rename : media/webrtc/trunk/webrtc/base/objc/RTCLogging.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCLogging.mm
rename : media/webrtc/trunk/webrtc/api/objc/RTCMediaConstraints+Private.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCMediaConstraints+Private.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCMediaConstraints.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCMediaConstraints.mm
rename : media/webrtc/trunk/webrtc/api/objc/RTCMediaSource.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCMediaSource.mm
rename : media/webrtc/trunk/webrtc/api/objc/RTCMediaStreamTrack+Private.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCMediaStreamTrack+Private.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCNSGLVideoView.m => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCNSGLVideoView.m
rename : media/webrtc/trunk/webrtc/api/objc/RTCOpenGLVideoRenderer.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCOpenGLVideoRenderer.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCSessionDescription+Private.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCSessionDescription+Private.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCSessionDescription.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCSessionDescription.mm
rename : media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.cc => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_decoder.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_decoder.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.cc => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_nalu.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_nalu.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu_unittest.cc => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_nalu_unittest.cc
rename : media/webrtc/trunk/webrtc/base/objc/RTCCameraPreviewView.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
rename : media/webrtc/trunk/webrtc/base/objc/RTCDispatcher.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCEAGLVideoView.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCIceCandidate.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCIceServer.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCStatsReport.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h
rename : media/webrtc/trunk/webrtc/base/objc/RTCLogging.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLogging.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCMediaConstraints.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCMediaSource.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCMediaStreamTrack.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCNSGLVideoView.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCSessionDescription.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCVideoFrame.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCVideoRenderer.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCMediaSource.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h
rename : media/webrtc/trunk/webrtc/api/objctests/RTCIceCandidateTest.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/UnitTests/RTCIceCandidateTest.mm
rename : media/webrtc/trunk/webrtc/api/objctests/RTCIceServerTest.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/UnitTests/RTCIceServerTest.mm
rename : media/webrtc/trunk/webrtc/api/objctests/RTCMediaConstraintsTest.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/UnitTests/RTCMediaConstraintsTest.mm
rename : media/webrtc/trunk/webrtc/api/objctests/RTCSessionDescriptionTest.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/UnitTests/RTCSessionDescriptionTest.mm
rename : media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_mac.cc => media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_darwin.cc
rename : media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_posix.cc => media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_non_darwin_unix.cc
rename : media/webrtc/trunk/webrtc/video/full_stack_plot.py => media/webrtc/trunk/webrtc/video/full_stack_tests_plot.py
rename : media/webrtc/trunk/webrtc/call/transport_adapter.cc => media/webrtc/trunk/webrtc/video/transport_adapter.cc
rename : media/webrtc/trunk/webrtc/call/transport_adapter.h => media/webrtc/trunk/webrtc/video/transport_adapter.h
rename : media/webrtc/trunk/webrtc/modules/utility/source/file_player_unittests.cc => media/webrtc/trunk/webrtc/voice_engine/file_player_unittests.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/channel_transport.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/channel_transport.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/channel_transport.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/channel_transport.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/traffic_control_win.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/traffic_control_win.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/traffic_control_win.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/traffic_control_win.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_manager_win.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket2_manager_win.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_manager_win.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket2_manager_win.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_win.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket2_win.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_win.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket2_win.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_posix.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_manager_posix.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_posix.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_manager_posix.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_unittest.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_manager_unittest.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_wrapper.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_manager_wrapper.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_wrapper.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_manager_wrapper.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_posix.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_posix.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_posix.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_posix.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_wrapper.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_wrapper.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_wrapper.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_wrapper.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_wrapper_unittest.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_wrapper_unittest.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_transport.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_transport.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_transport_impl.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_transport_impl.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_transport_impl.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_transport_impl.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_transport_unittest.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_transport_unittest.cc
2017-06-13 01:52:22 -04:00
a39099bca1
Bug 1166955 - Stop including nsAutoPtr.h from BasePin.cpp. r=jesup
...
MozReview-Commit-ID: CCvJyVmH1JI
--HG--
extra : rebase_source : 72cf22b38f7c4d3ef5c315351f0eca942345b752
2017-06-03 17:59:29 +09:00
858e777c78
Bug 1369108 - 3. Implement new device permission code path for Fennec; r=esawin
...
Instead of asking for permission in VideoCaptureDeviceInfoAndroid.java,
we now merely check for permission there. The actual permission prompt
now happens in WebrtcUI.js, using the new
"getUserMedia:ask-device-permission" and
"getUserMedia:got-device-permission" notifications.
MozReview-Commit-ID: DSVPjjW2JNR
2017-06-02 16:11:53 -04:00
9798dcedbb
Bug 1369108 - 2. Refresh Android camera list when necessary; r=jesup
...
Currently, if permission is first denied, the list of cameras is empty.
However, if permission is later granted, the list stays empty because we
never try to refresh the list. This patch causes us to refresh the list
when necessary.
MozReview-Commit-ID: 5eodPCWVyaP
2017-06-02 16:11:53 -04:00
e7f79e098e
Bug 1363885 - 2. Remove ViERenderer dependency on orientation listener; r=snorp
...
ViERenderer is not used anywhere but has a couple calls to the obsolete
GeckoAppShell orientation listener. The entire ViERenderer.java file is
getting removed in the upcoming WebRTC update, so we should just go
ahead and remove those lines.
MozReview-Commit-ID: AwG7dBg5MV8
2017-05-25 18:33:30 -04:00
4dd7f2dc11
Bug 1365090: use target bitrate instead of max for simulcast. r=bwc
...
MozReview-Commit-ID: GThcXHHnoCV
--HG--
extra : rebase_source : 352a82ad81858782898a10440ff77b4891af6a60
2017-05-16 16:15:04 -07:00
6856ba9b3d
Bug 1363259 - set min and max fps through AVCaptureConnection. r=jib
...
MozReview-Commit-ID: 4GY1gOICLqU
--HG--
extra : rebase_source : 97b50fa34186f9c92f0d01e1d486137b5159a8bd
2017-05-09 10:28:31 +08:00
4e635ff251
Bug 1358224 - pt 3 - fix leak in RTPHeaderExtension's rid char buffer. r=drno
...
Turns out since Firefox doesn't receive simulcast streams, we never
noticed this leak. Convert RTPHeaderExtension.rid from a char* to
rtc::scoped_ptr<char[]> so it gets deleted properly. This also
requires a new copy constructor and assignment operator.
MozReview-Commit-ID: Jh4Gp4dAl9g
--HG--
extra : rebase_source : 8c1081fecd6e56a8f932af54fbd294adb85866f5
2017-04-27 12:27:02 -05:00
4b1556a5f2
Bug 1355056 - replace (function(args) { /* do stuff using this */ }).bind(this) with arrow functions, r=jaws.
2017-04-27 00:25:45 +02:00
514e230373
Merge inbound to central, a=merge
2017-04-13 17:24:01 -07:00
a9c52a60b1
Bug 1241066 - fix mozRtt always 0 or 1;r=jib
...
My shortest patch to date.
MozReview-Commit-ID: 8r3ZrGUk40D
--HG--
extra : rebase_source : 38cc51ce85e03c03f46e063bf92f594927d1365f
2017-03-20 16:58:53 -07:00
75a0220f53
Bug 1325513: Check RTP extension header length. r=jesup
...
MozReview-Commit-ID: 6sUVQjUh8bF
--HG--
extra : rebase_source : 296cb8688a9c27b437380e5f70fd3cf9d43629f2
2017-04-12 15:09:18 -07:00
af67a2fb4c
Bug 1348657 - implement framesEncoded, pliCount, nackCount and firCount for webrtc stats r=jib,smaug
...
MozReview-Commit-ID: E873mbcrlLL
--HG--
extra : rebase_source : ca6f5d7ab0490948aaed1ae793ed5906149b7236
2017-03-21 21:52:06 -07:00
d5d480e33d
Bug 1349581: defer nativeRegistration for android Jni to avoid thread issues r=gcp
...
MozReview-Commit-ID: Ep0ej5HkGE3
2017-04-03 16:58:44 -04:00
1d80aba420
Bug 1351700 - Fix linking error when building webrtc-gtest; r=jesup
...
This adds a stub implementation for nsTraceRefcnt::WalkTheStack which we're
pulling in from <mozilla/Assertions.h> in some debug builds.
MozReview-Commit-ID: 6wVghIfKWWZ
--HG--
extra : rebase_source : 1e8472935c7f8ac486794fab764e08b30eea79ed
2017-03-29 14:22:23 -04:00
774e2f6945
Bug 964133 - Build webrtc.org unit tests; r=jesup,ted.mielczarik
...
This adds a moz.build file for the tests. The alternative would be to hack up
the gyp files. Since gyp support has already been removed from upstream, this
does not really buy us anything as far as maintainabily goes. Once gn support
is available in our build system, we can remove this moz.build file and use
the gn files instead.
The include paths for the gtest and gmock headers in the webrtc.org tests are
not compatible with where we export the headers. We could patch each unittest,
but the include location has already changed upstream making this painful
to maintain. Instead, we duplicate the relevant headers to the expected path.
MozReview-Commit-ID: 1ADUAMxTCFq
--HG--
extra : rebase_source : 2cc10faa7018ee8af8e8f3d7805265ed2dd89507
2017-03-27 15:51:16 -04:00
164d41ff3a
Bug 964133 - Add stub implementation of OSXRunLoopSingleton.cpp; r=jesup
...
MozReview-Commit-ID: 8yJh0V2rLoR
--HG--
extra : rebase_source : 3df843511ecb40af2b6d83030453758e838e94be
2017-02-07 13:47:00 -05:00
f44831c17d
Bug 964133 - Build gflags; r=ted.mielczarik
...
This adds gflags to the list of ignored directories for clang static
analysis and adds "explicit" where required in mutex.h.
We also stop building a duplicate copy of snprintf for windows as our builds
already include a definition for it.
MozReview-Commit-ID: 4uMhTMvAKL0
--HG--
extra : rebase_source : d63d3797053c7720c725b3994cb3b2ca11bb191f
2017-03-28 15:46:57 -04:00
15ed133076
Bug 964133 - Import gflags from webrtc.org branch 49; r=ted.mielczarik
...
MozReview-Commit-ID: 2AouGZrZa9w
--HG--
extra : rebase_source : 76ab8e4a280912aa08a6c94e1cdfb25488df836a
2017-03-08 11:33:01 -05:00
6affa752e1
Bug 964133 - Fixup FakeIPC to build on windows; r=jesup
...
MozReview-Commit-ID: 2Jm82qs4hFL
--HG--
extra : rebase_source : dfade0944c84df23580121384d2501229b78c0c6
2017-01-30 15:37:57 -05:00
d71edf00c9
Bug 964133 - Move FakeIPC to webrtc.org gtest; r=jesup
...
These were written for the signaling tests but are no longer needed there
because those tests link against libxul. They are still useful for the
webrtc.org unit tests, so we should move them there.
MozReview-Commit-ID: GVskiZebq19
--HG--
rename : media/webrtc/signaling/test/FakeIPC.cpp => media/webrtc/trunk/gtest/FakeIPC.cpp
rename : media/webrtc/signaling/test/FakeIPC.h => media/webrtc/trunk/gtest/FakeIPC.h
extra : rebase_source : 57b70f5dd3a55e73de0b066f228ddf957f477c26
2017-01-13 14:03:17 -05:00
4205129c92
Bug 964133 - Remove webrtc.org copy of gtest; r=ted.mielczarek
...
MozReview-Commit-ID: LURWNU2zwRT
--HG--
extra : rebase_source : 5a0c8cd1e56bfbe82e581bce9dcf2cb126a605c5
2017-03-28 15:44:38 -04:00
e7dd39ae18
Bug 1347813: take cumulativeLost from RTCP. r=jesup
...
MozReview-Commit-ID: LIPxL8vcnHl
--HG--
extra : rebase_source : e7df61a4200e088dba0b5f85d59669e99d35476f
2017-03-20 14:03:46 -07:00
194702d7ab
Bug 1343691 - fix missing rtcp stats;r=jib
...
Omitting the RTT when it is not available breaks a lot of tests (as jesup warned).
I am going to fix the RTT behavior and the tests in bug 1344970, for now RTT will
be zero when unavailable.
MozReview-Commit-ID: 9x3eQfbM3ZT
--HG--
extra : rebase_source : f8d46d7232455a3038fd99ffb6cc14111c44a794
2017-03-08 23:26:24 -08:00
f587b4fa3f
Bug 1325173 - read full RtpStreamId when parsing RTP header extensions. r=drno
...
MozReview-Commit-ID: CHkqA0MM3fx
--HG--
extra : rebase_source : 84c0e85c9f214f1bc7403256d8c2d80809305e13
2017-03-05 23:37:51 -08:00
e84fc624ff
Merge inbound to central, a=merge
...
MozReview-Commit-ID: DpCZgRV1csS
2017-02-24 16:46:12 -08:00
8ad25a673c
Bug 1284800: Fix build fallout from moving libyuv into a subdirectory r=ted
...
MozReview-Commit-ID: CDMDXqpGueS
2017-02-24 14:01:56 -05:00
47dc9207cd
Backed out changeset 20a81b2adf80 (bug 1330240) under suspicion of turning android mda1 nearly permafail a=backout
...
MozReview-Commit-ID: LUKhxorIzwU
2017-02-23 16:25:18 -08:00
71b7ced622
Bug 1332622 - Remove MOZILLA_INTERNAL_API macro from webrtc; r=jesup
...
I've also cleaned up a few leftover references to USE_FAKE_MEDIA_STREAMS,
MOZILLA_EXTERNAL_LINKAGE and MOZ_WIDGET_GONK where I noticed them.
MozReview-Commit-ID: Cdo1Y4IrFqp
--HG--
extra : rebase_source : 4f7debb5ebc3e024410ec6456fae0d3463ca1d10
2017-02-22 09:07:32 -05:00
959d5215d6
Bug 1330240 - Limit -Wthread-safety to WebRTC due to lack of annotations. r=cpeterson,froydnj,jesup
...
MozReview-Commit-ID: HuoXFwZkdYo
--HG--
extra : rebase_source : 8f07a7a6de6d794b26b0f2b18eb95452d65c8f40
2017-01-11 16:50:18 +00:00
b2013667fb
Bug 1330240 - Limit -Wthread-safety to WebRTC due to lack of annotations. r=cpeterson,froydnj,jesup
...
MozReview-Commit-ID: HuoXFwZkdYo
--HG--
extra : rebase_source : 8f07a7a6de6d794b26b0f2b18eb95452d65c8f40
2017-01-11 16:50:18 +00:00
76acc650ab
Bug 1301286: backout accidental logging change rs=backout
...
MozReview-Commit-ID: C2V6McYpfh0
2017-02-16 21:29:28 -05:00
39cdece7fe
Bug 1300665: Add abs-send-time and toffset header extension usage and negotiation r=bwc
...
MozReview-Commit-ID: 3h9C8XziNky
2017-02-09 20:56:29 -05:00
729ec22dc5
Bug 1301286: At least in the webrtc49 update, 100Kbps isn't enough for simulcast tests r=abr
...
MozReview-Commit-ID: kQHNnr7rAg
2017-02-16 15:37:03 -05:00
33173f619b
Backed out changeset 2a2ffd6f443c (bug 1300665) for failures in test_peerConnection_simulcastOffer.html
...
CLOSED TREE
2017-02-13 18:55:31 -08:00
c8fa3242ab
Merge m-c to inbound, a=merge
...
MozReview-Commit-ID: Lt0WpWkto4h
2017-02-13 17:07:33 -08:00
30a3f082fd
Bug 1337468: pass RID values via RTP configuration r=ng
...
MozReview-Commit-ID: Gl5TdZkJIZ8
--HG--
extra : rebase_source : 296d74fcfee7535b052e8f97ba65f0c67afbb129
2017-02-08 21:37:13 -08:00
d03b7e603e
Bug 1337468: removed unused RID code and variables r=ng
...
MozReview-Commit-ID: JWBRVC7WQsl
--HG--
extra : rebase_source : 0846deda23642804dcbdfcf078e5d7e0e0ee4bd1
2017-02-08 21:27:32 -08:00
eb37fd7ca3
Bug 1338000 - Fix webrtc build on non-Android ARM Linux. r=jesup
...
--HG--
extra : rebase_source : a6556bfa40a1b98506d61cb056dde3bb6313c559
2017-02-09 10:26:01 +09:00
db244f7210
Bug 1332664: Cherry-pick upstream webrtc.org patch to not depend on 'experimental' includes for libvpx r=rillian
2017-01-27 21:10:58 -05:00
96cc4073b0
Bug 1325299 - Don't explicitly set PSAPI_VERSION. r=glandium
...
WINVER=0x0601 implies PSAPI_VERSION=2. We should not mix PSAPI_VERSION.
MozReview-Commit-ID: Ckxel4JNW2x
--HG--
extra : rebase_source : 3dc221ca67642ea810cb353869f76b82c40c7bf3
2016-12-30 01:29:52 +09:00
a50658a789
Bug 1331648 - detect device connection for the case /dev/v4l hasn't existed; r=jesup
...
MozReview-Commit-ID: 1erqrFScjr
--HG--
extra : rebase_source : ad4a4adde64f479a84a3f7fa093f71eef319436b
2017-01-25 14:52:19 +08:00
19cb0f9bd3
Bug 1333752: Fix memset sizes in rtt_filter.cc r=dminor
...
MozReview-Commit-ID: 3pnTseY4k2M
2017-01-25 13:28:13 -05:00
aee89e79af
Bug 1300665: Add abs-send-time and toffset header extension usage and negotiation r=bwc
...
MozReview-Commit-ID: 3h9C8XziNky
2017-02-09 20:56:29 -05:00
8e94f85a7d
Bug 1339270: Add rtp 'padding' packets into rtp history for handling NACKs r=ng
...
Webrtc.org issue 7143:
https://bugs.chromium.org/p/webrtc/issues/detail?id=7143
MozReview-Commit-ID: 9pGj63gWSC6
2017-02-13 18:44:15 -05:00
4d6ebaaa07
Bug 1332139: make system changes to fix libvpx include paths (prefer media/libvpx) r=ted
2017-01-20 10:42:32 -05:00
33fe82dde6
Bug 1332139: Remove LIBVPX_SVC hack for vp9 needed to work with libvpx 1.4 from webrtc r=ng
2017-01-20 10:42:30 -05:00
85ae466ed9
Bug 1329194 - media:webrtc: Add platform defines for SH. r=jesup
2017-01-20 09:12:03 +09:00
187beffa39
Merge m-c to autoland. a=merge
...
--HG--
extra : rebase_source : 0de29cc9f544d8882d3e8c13572d3c4b98ba3c26
2017-01-18 09:59:53 -05:00
ca0bc14941
Bug 1329922 - Change kMinTelephoneEventDuration to match minimum duration in WebRTC spec; r=jesup
...
The spec says 40 is the minimum, but here it is set to 100.
MozReview-Commit-ID: AUy7AviiYVh
--HG--
extra : rebase_source : bbebb4fba9a5f853e4d041029ad02820a4d564ce
2017-01-17 09:59:11 -05:00
5262f0e0a9
Bug 1331158: Install new receive codec config for WebRTC if it changed r=ng
...
Adds a missing API parameter to upstream webrtc.org code in the Call API
2017-01-17 15:50:14 -05:00
9fd9034ad9
Bug 1275204 - media:webrtc: Use better pre-processor defines for sparc64. r=jesup
2017-01-11 16:09:44 +09:00
2663bd16e0
Bug 1326463 - Fix OpenBSD build broken by webrtc.org 49 update. r=gaston, r=jesup
...
--HG--
extra : histedit_source : e18f813ace63db9f9ea6b35437e9b87bb84f4e26%2C9da31e71e8591e1ba49000c0261b55404802b32d
2017-01-06 11:36:00 -05:00
abfe13cbc5
Bug 1326288: Restore patches for bug 1237023 and bug 1315283 - lost in 49 update r=pkerr
2016-12-30 00:11:59 -05:00
a1d4774bcd
Bug 1326011 - Unbreak build on BSDs after after bug 1250356. r=jesup
...
MozReview-Commit-ID: 390dsKqlJQe
2016-12-28 00:57:00 +01:00
ac570b16c4
Bug 1250356: rollup of changes for webrtc after applying webrtc.org v49 update r=pkerr,ng,pehrsons,etc
...
See ssh://hg.mozilla.org/users/paulrkerr_gmail.com/webrtc49_merge/ for the
patch development history.
2016-12-27 19:41:02 -05:00
5b52deb57a
Bug 1250356: Webrtc updated to upstream branch 49; pull made 2016-02-22 by pkerr rs=jesup
...
--HG--
rename : media/webrtc/trunk/webrtc/modules/video_processing/main/source/OWNERS => media/webrtc/trunk/webrtc/audio/OWNERS
rename : media/webrtc/trunk/webrtc/video_engine/OWNERS => media/webrtc/trunk/webrtc/call/OWNERS
rename : media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_allocator.cc => media/webrtc/trunk/webrtc/call/bitrate_allocator.cc
rename : media/webrtc/trunk/webrtc/modules/bitrate_controller/include/bitrate_allocator.h => media/webrtc/trunk/webrtc/call/bitrate_allocator.h
rename : media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_allocator_unittest.cc => media/webrtc/trunk/webrtc/call/bitrate_allocator_unittest.cc
rename : media/webrtc/trunk/webrtc/video/call_perf_tests.cc => media/webrtc/trunk/webrtc/call/call_perf_tests.cc
rename : media/webrtc/trunk/webrtc/common_video/interface/i420_buffer_pool.h => media/webrtc/trunk/webrtc/common_video/include/i420_buffer_pool.h
rename : media/webrtc/trunk/webrtc/common_video/interface/video_image.h => media/webrtc/trunk/webrtc/common_video/include/video_image.h
rename : media/webrtc/trunk/webrtc/modules/video_render/video_render_frames.cc => media/webrtc/trunk/webrtc/common_video/video_render_frames.cc
rename : media/webrtc/trunk/webrtc/modules/video_render/video_render_frames.h => media/webrtc/trunk/webrtc/common_video/video_render_frames.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/audio_codec_speed_tests.isolate => media/webrtc/trunk/webrtc/modules/audio_codec_speed_tests.isolate
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_neteq_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_neteq_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_resampler.h => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_resampler.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/call_statistics.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/call_statistics.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/call_statistics.h => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/call_statistics.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/call_statistics_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/call_statistics_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/initial_delay_manager.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/initial_delay_manager.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h => media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h => media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/g711_interface.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h => media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_interface.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h => media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/ilbc.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h => media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h => media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/include/isac.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h => media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_interface.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h => media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq/interface/neteq.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/include/neteq.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/nack.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/nack.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/ACMTest.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/ACMTest.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/APITest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/APITest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/APITest.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/APITest.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/Channel.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/Channel.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/Channel.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/Channel.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/EncodeDecodeTest.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PCMFile.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/PCMFile.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PCMFile.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/PCMFile.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PacketLossTest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/PacketLossTest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PacketLossTest.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/PacketLossTest.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/RTPFile.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/RTPFile.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/RTPFile.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/RTPFile.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/SpatialAudio.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/SpatialAudio.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/TestAllCodecs.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestAllCodecs.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/TestAllCodecs.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestRedFec.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/TestRedFec.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestRedFec.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/TestRedFec.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestStereo.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/TestStereo.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestStereo.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/TestStereo.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestVADDTX.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/TestVADDTX.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestVADDTX.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/TestVADDTX.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/Tester.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/Tester.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TimedTrace.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/TimedTrace.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TimedTrace.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/TimedTrace.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TwoWayCommunication.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/TwoWayCommunication.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/delay_test.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/delay_test.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/iSACTest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/iSACTest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/iSACTest.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/iSACTest.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/opus_test.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/opus_test.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/target_delay_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/utility.cc => media/webrtc/trunk/webrtc/modules/audio_coding/test/utility.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/utility.h => media/webrtc/trunk/webrtc/modules/audio_coding/test/utility.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_unittests.isolate => media/webrtc/trunk/webrtc/modules/audio_decoder_unittests.isolate
rename : media/webrtc/trunk/webrtc/modules/audio_device/android/fine_audio_buffer_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_device/audio_device_tests.isolate => media/webrtc/trunk/webrtc/modules/audio_device_tests.isolate
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aec/include/echo_cancellation.h => media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation.h
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h => media/webrtc/trunk/webrtc/modules/audio_processing/aecm/echo_control_mobile.h
rename : media/webrtc/trunk/webrtc/modules/audio_processing/ns/include/noise_suppression.h => media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression.h
rename : media/webrtc/trunk/webrtc/modules/audio_processing/ns/include/noise_suppression_x.h => media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression_x.h
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/common.h => media/webrtc/trunk/webrtc/modules/audio_processing/vad/common.h
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/gmm.cc => media/webrtc/trunk/webrtc/modules/audio_processing/vad/gmm.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/gmm.h => media/webrtc/trunk/webrtc/modules/audio_processing/vad/gmm.h
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/gmm_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_processing/vad/gmm_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_based_vad.cc => media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_based_vad.h => media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_based_vad.h
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_based_vad_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_based_vad_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_internal.cc => media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_internal.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_internal.h => media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_internal.h
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_internal_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_internal_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/standalone_vad.cc => media/webrtc/trunk/webrtc/modules/audio_processing/vad/standalone_vad.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/standalone_vad.h => media/webrtc/trunk/webrtc/modules/audio_processing/vad/standalone_vad.h
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/standalone_vad_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_audio_proc_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/circular_buffer.cc => media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_circular_buffer.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/circular_buffer.h => media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_circular_buffer.h
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/circular_buffer_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_circular_buffer_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/interface/module.h => media/webrtc/trunk/webrtc/modules/include/module.h
rename : media/webrtc/trunk/webrtc/modules/interface/module_common_types.h => media/webrtc/trunk/webrtc/modules/include/module_common_types.h
rename : media/webrtc/trunk/webrtc/modules/media_file/interface/media_file.h => media/webrtc/trunk/webrtc/modules/media_file/media_file.h
rename : media/webrtc/trunk/webrtc/modules/media_file/interface/media_file_defines.h => media/webrtc/trunk/webrtc/modules/media_file/media_file_defines.h
rename : media/webrtc/trunk/webrtc/modules/media_file/source/media_file_impl.cc => media/webrtc/trunk/webrtc/modules/media_file/media_file_impl.cc
rename : media/webrtc/trunk/webrtc/modules/media_file/source/media_file_impl.h => media/webrtc/trunk/webrtc/modules/media_file/media_file_impl.h
rename : media/webrtc/trunk/webrtc/modules/media_file/source/media_file_unittest.cc => media/webrtc/trunk/webrtc/modules/media_file/media_file_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/media_file/source/media_file_utility.cc => media/webrtc/trunk/webrtc/modules/media_file/media_file_utility.cc
rename : media/webrtc/trunk/webrtc/modules/media_file/source/media_file_utility.h => media/webrtc/trunk/webrtc/modules/media_file/media_file_utility.h
rename : media/webrtc/trunk/webrtc/modules/pacing/include/mock/mock_paced_sender.h => media/webrtc/trunk/webrtc/modules/pacing/mock/mock_paced_sender.h
rename : media/webrtc/trunk/webrtc/modules/pacing/include/paced_sender.h => media/webrtc/trunk/webrtc/modules/pacing/paced_sender.h
rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/fec_receiver.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/fec_receiver.h
rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/receive_statistics.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/receive_statistics.h
rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_cvo.h
rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h
rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
rename : media/webrtc/trunk/webrtc/modules/utility/interface/audio_frame_operations.h => media/webrtc/trunk/webrtc/modules/utility/include/audio_frame_operations.h
rename : media/webrtc/trunk/webrtc/modules/utility/interface/file_player.h => media/webrtc/trunk/webrtc/modules/utility/include/file_player.h
rename : media/webrtc/trunk/webrtc/modules/utility/interface/file_recorder.h => media/webrtc/trunk/webrtc/modules/utility/include/file_recorder.h
rename : media/webrtc/trunk/webrtc/modules/utility/interface/mock/mock_process_thread.h => media/webrtc/trunk/webrtc/modules/utility/include/mock/mock_process_thread.h
rename : media/webrtc/trunk/webrtc/modules/utility/interface/process_thread.h => media/webrtc/trunk/webrtc/modules/utility/include/process_thread.h
rename : media/webrtc/trunk/webrtc/modules/video_capture/include/video_capture.h => media/webrtc/trunk/webrtc/modules/video_capture/video_capture.h
rename : media/webrtc/trunk/webrtc/modules/video_capture/include/video_capture_defines.h => media/webrtc/trunk/webrtc/modules/video_capture/video_capture_defines.h
rename : media/webrtc/trunk/webrtc/modules/video_capture/include/video_capture_factory.h => media/webrtc/trunk/webrtc/modules/video_capture/video_capture_factory.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/codec_database.h => media/webrtc/trunk/webrtc/modules/video_coding/codec_database.h
rename : media/webrtc/trunk/webrtc/modules/video_processing/main/source/OWNERS => media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/OWNERS
rename : media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/main/source/i420.cc => media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/i420.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/main/source/i420.gypi => media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/i420.gypi
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/content_metrics_processing.h => media/webrtc/trunk/webrtc/modules/video_coding/content_metrics_processing.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/decoding_state.cc => media/webrtc/trunk/webrtc/modules/video_coding/decoding_state.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/decoding_state.h => media/webrtc/trunk/webrtc/modules/video_coding/decoding_state.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc => media/webrtc/trunk/webrtc/modules/video_coding/decoding_state_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/encoded_frame.cc => media/webrtc/trunk/webrtc/modules/video_coding/encoded_frame.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/frame_buffer.h => media/webrtc/trunk/webrtc/modules/video_coding/frame_buffer.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_buffer.cc => media/webrtc/trunk/webrtc/modules/video_coding/jitter_buffer.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_buffer.h => media/webrtc/trunk/webrtc/modules/video_coding/jitter_buffer.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_estimator_tests.cc => media/webrtc/trunk/webrtc/modules/video_coding/jitter_estimator_tests.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/media_optimization.cc => media/webrtc/trunk/webrtc/modules/video_coding/media_optimization.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/media_optimization.h => media/webrtc/trunk/webrtc/modules/video_coding/media_optimization.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/qm_select.cc => media/webrtc/trunk/webrtc/modules/video_coding/qm_select.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/qm_select.h => media/webrtc/trunk/webrtc/modules/video_coding/qm_select.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/qm_select_unittest.cc => media/webrtc/trunk/webrtc/modules/video_coding/qm_select_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/receiver.cc => media/webrtc/trunk/webrtc/modules/video_coding/receiver.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/session_info.cc => media/webrtc/trunk/webrtc/modules/video_coding/session_info.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/session_info.h => media/webrtc/trunk/webrtc/modules/video_coding/session_info.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/session_info_unittest.cc => media/webrtc/trunk/webrtc/modules/video_coding/session_info_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/test/plotJitterEstimate.m => media/webrtc/trunk/webrtc/modules/video_coding/test/plotJitterEstimate.m
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/test/plotReceiveTrace.m => media/webrtc/trunk/webrtc/modules/video_coding/test/plotReceiveTrace.m
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/test/plotTimingTest.m => media/webrtc/trunk/webrtc/modules/video_coding/test/plotTimingTest.m
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/test/release_test.h => media/webrtc/trunk/webrtc/modules/video_coding/test/release_test.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/test/rtp_player.cc => media/webrtc/trunk/webrtc/modules/video_coding/test/rtp_player.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/test/rtp_player.h => media/webrtc/trunk/webrtc/modules/video_coding/test/rtp_player.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/test/stream_generator.cc => media/webrtc/trunk/webrtc/modules/video_coding/test/stream_generator.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/test/subfigure.m => media/webrtc/trunk/webrtc/modules/video_coding/test/subfigure.m
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/test/test_util.cc => media/webrtc/trunk/webrtc/modules/video_coding/test/test_util.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/test/test_util.h => media/webrtc/trunk/webrtc/modules/video_coding/test/test_util.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc => media/webrtc/trunk/webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.h => media/webrtc/trunk/webrtc/modules/video_coding/test/vcm_payload_sink_factory.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/test/video_rtp_play.cc => media/webrtc/trunk/webrtc/modules/video_coding/test/video_rtp_play.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/timing.cc => media/webrtc/trunk/webrtc/modules/video_coding/timing.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/timing.h => media/webrtc/trunk/webrtc/modules/video_coding/timing.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/timing_unittest.cc => media/webrtc/trunk/webrtc/modules/video_coding/timing_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_coding_impl.h => media/webrtc/trunk/webrtc/modules/video_coding/video_coding_impl.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_receiver_unittest.cc => media/webrtc/trunk/webrtc/modules/video_coding/video_receiver_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_sender_unittest.cc => media/webrtc/trunk/webrtc/modules/video_coding/video_sender_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/video_processing/main/source/content_analysis.h => media/webrtc/trunk/webrtc/modules/video_processing/content_analysis.h
rename : media/webrtc/trunk/webrtc/modules/video_processing/main/source/deflickering.cc => media/webrtc/trunk/webrtc/modules/video_processing/deflickering.cc
rename : media/webrtc/trunk/webrtc/modules/video_processing/main/source/spatial_resampler.cc => media/webrtc/trunk/webrtc/modules/video_processing/spatial_resampler.cc
rename : media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/createTable.m => media/webrtc/trunk/webrtc/modules/video_processing/test/createTable.m
rename : media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/readYUV420file.m => media/webrtc/trunk/webrtc/modules/video_processing/test/readYUV420file.m
rename : media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/writeYUV420file.m => media/webrtc/trunk/webrtc/modules/video_processing/test/writeYUV420file.m
rename : media/webrtc/trunk/webrtc/modules/video_processing/main/source/video_decimator.cc => media/webrtc/trunk/webrtc/modules/video_processing/video_decimator.cc
rename : media/webrtc/trunk/webrtc/modules/video_processing/main/source/video_decimator.h => media/webrtc/trunk/webrtc/modules/video_processing/video_decimator.h
rename : media/webrtc/trunk/webrtc/modules/video_render/include/video_render.h => media/webrtc/trunk/webrtc/modules/video_render/video_render.h
rename : media/webrtc/trunk/webrtc/modules/video_render/include/video_render_defines.h => media/webrtc/trunk/webrtc/modules/video_render/video_render_defines.h
rename : media/webrtc/trunk/webrtc/modules/video_render/video_render_tests.isolate => media/webrtc/trunk/webrtc/modules/video_render_tests.isolate
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/aligned_malloc.h => media/webrtc/trunk/webrtc/system_wrappers/include/aligned_malloc.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/asm_defines.h => media/webrtc/trunk/webrtc/system_wrappers/include/asm_defines.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/atomic32.h => media/webrtc/trunk/webrtc/system_wrappers/include/atomic32.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/clock.h => media/webrtc/trunk/webrtc/system_wrappers/include/clock.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/compile_assert_c.h => media/webrtc/trunk/webrtc/system_wrappers/include/compile_assert_c.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/condition_variable_wrapper.h => media/webrtc/trunk/webrtc/system_wrappers/include/condition_variable_wrapper.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/cpu_features_wrapper.h => media/webrtc/trunk/webrtc/system_wrappers/include/cpu_features_wrapper.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/cpu_info.h => media/webrtc/trunk/webrtc/system_wrappers/include/cpu_info.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/critical_section_wrapper.h => media/webrtc/trunk/webrtc/system_wrappers/include/critical_section_wrapper.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/data_log.h => media/webrtc/trunk/webrtc/system_wrappers/include/data_log.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/data_log_c.h => media/webrtc/trunk/webrtc/system_wrappers/include/data_log_c.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/data_log_impl.h => media/webrtc/trunk/webrtc/system_wrappers/include/data_log_impl.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/event_wrapper.h => media/webrtc/trunk/webrtc/system_wrappers/include/event_wrapper.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/field_trial.h => media/webrtc/trunk/webrtc/system_wrappers/include/field_trial.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/field_trial_default.h => media/webrtc/trunk/webrtc/system_wrappers/include/field_trial_default.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/file_wrapper.h => media/webrtc/trunk/webrtc/system_wrappers/include/file_wrapper.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/fix_interlocked_exchange_pointer_win.h => media/webrtc/trunk/webrtc/system_wrappers/include/fix_interlocked_exchange_pointer_win.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/logcat_trace_context.h => media/webrtc/trunk/webrtc/system_wrappers/include/logcat_trace_context.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/logging.h => media/webrtc/trunk/webrtc/system_wrappers/include/logging.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/ref_count.h => media/webrtc/trunk/webrtc/system_wrappers/include/ref_count.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/rtp_to_ntp.h => media/webrtc/trunk/webrtc/system_wrappers/include/rtp_to_ntp.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/rw_lock_wrapper.h => media/webrtc/trunk/webrtc/system_wrappers/include/rw_lock_wrapper.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/scoped_vector.h => media/webrtc/trunk/webrtc/system_wrappers/include/scoped_vector.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/sleep.h => media/webrtc/trunk/webrtc/system_wrappers/include/sleep.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/sort.h => media/webrtc/trunk/webrtc/system_wrappers/include/sort.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/stl_util.h => media/webrtc/trunk/webrtc/system_wrappers/include/stl_util.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/stringize_macros.h => media/webrtc/trunk/webrtc/system_wrappers/include/stringize_macros.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/tick_util.h => media/webrtc/trunk/webrtc/system_wrappers/include/tick_util.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/timestamp_extrapolator.h => media/webrtc/trunk/webrtc/system_wrappers/include/timestamp_extrapolator.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/trace.h => media/webrtc/trunk/webrtc/system_wrappers/include/trace.h
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/utf_util_win.h => media/webrtc/trunk/webrtc/system_wrappers/include/utf_util_win.h
rename : media/webrtc/trunk/webrtc/system_wrappers/source/event_posix.cc => media/webrtc/trunk/webrtc/system_wrappers/source/event_timer_posix.cc
rename : media/webrtc/trunk/webrtc/system_wrappers/source/event_posix.h => media/webrtc/trunk/webrtc/system_wrappers/source/event_timer_posix.h
rename : media/webrtc/trunk/webrtc/system_wrappers/source/event_win.h => media/webrtc/trunk/webrtc/system_wrappers/source/event_timer_win.h
rename : media/webrtc/trunk/webrtc/video_engine/call_stats.cc => media/webrtc/trunk/webrtc/video/call_stats.cc
rename : media/webrtc/trunk/webrtc/video_engine/call_stats.h => media/webrtc/trunk/webrtc/video/call_stats.h
rename : media/webrtc/trunk/webrtc/video_engine/call_stats_unittest.cc => media/webrtc/trunk/webrtc/video/call_stats_unittest.cc
rename : media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback.cc => media/webrtc/trunk/webrtc/video/encoder_state_feedback.cc
rename : media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback.h => media/webrtc/trunk/webrtc/video/encoder_state_feedback.h
rename : media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback_unittest.cc => media/webrtc/trunk/webrtc/video/encoder_state_feedback_unittest.cc
rename : media/webrtc/trunk/webrtc/video_engine/payload_router.cc => media/webrtc/trunk/webrtc/video/payload_router.cc
rename : media/webrtc/trunk/webrtc/video_engine/payload_router.h => media/webrtc/trunk/webrtc/video/payload_router.h
rename : media/webrtc/trunk/webrtc/video_engine/payload_router_unittest.cc => media/webrtc/trunk/webrtc/video/payload_router_unittest.cc
rename : media/webrtc/trunk/webrtc/video_engine/report_block_stats.cc => media/webrtc/trunk/webrtc/video/report_block_stats.cc
rename : media/webrtc/trunk/webrtc/video_engine/report_block_stats.h => media/webrtc/trunk/webrtc/video/report_block_stats.h
rename : media/webrtc/trunk/webrtc/video_engine/report_block_stats_unittest.cc => media/webrtc/trunk/webrtc/video/report_block_stats_unittest.cc
rename : media/webrtc/trunk/webrtc/video_engine/stream_synchronization.cc => media/webrtc/trunk/webrtc/video/stream_synchronization.cc
rename : media/webrtc/trunk/webrtc/video_engine/stream_synchronization.h => media/webrtc/trunk/webrtc/video/stream_synchronization.h
rename : media/webrtc/trunk/webrtc/video_engine/stream_synchronization_unittest.cc => media/webrtc/trunk/webrtc/video/stream_synchronization_unittest.cc
rename : media/webrtc/trunk/webrtc/video_engine/vie_codec_unittest.cc => media/webrtc/trunk/webrtc/video/vie_codec_unittest.cc
rename : media/webrtc/trunk/webrtc/video_engine/vie_receiver.cc => media/webrtc/trunk/webrtc/video/vie_receiver.cc
rename : media/webrtc/trunk/webrtc/video_engine/vie_receiver.h => media/webrtc/trunk/webrtc/video/vie_receiver.h
rename : media/webrtc/trunk/webrtc/video_engine/vie_remb.cc => media/webrtc/trunk/webrtc/video/vie_remb.cc
rename : media/webrtc/trunk/webrtc/video_engine/vie_remb.h => media/webrtc/trunk/webrtc/video/vie_remb.h
rename : media/webrtc/trunk/webrtc/video_engine/vie_remb_unittest.cc => media/webrtc/trunk/webrtc/video/vie_remb_unittest.cc
rename : media/webrtc/trunk/webrtc/video_engine/vie_sync_module.cc => media/webrtc/trunk/webrtc/video/vie_sync_module.cc
rename : media/webrtc/trunk/webrtc/video_engine/vie_sync_module.h => media/webrtc/trunk/webrtc/video/vie_sync_module.h
rename : media/webrtc/trunk/webrtc/voice_engine/include/mock/mock_voe_observer.h => media/webrtc/trunk/webrtc/voice_engine/mock/mock_voe_observer.h
2016-12-27 19:40:48 -05:00
e551a0b979
Bug 1322465 part 7 - Use explicit/MOZ_IMPLICIT for the unary constructors in media/. r=cpearce,Ehsan
...
MozReview-Commit-ID: Ln63tzmkynd
--HG--
extra : rebase_source : 869cca55da15d2394411571f741d8ed2728c3265
2016-12-16 15:56:40 +08:00
fa49fdf95c
Bug 1324608: restrict RID len. r=ng
...
MozReview-Commit-ID: 9iA0s4EmGow
--HG--
extra : rebase_source : d9888dc04047d70968658e9257100cd654657550
2016-12-20 10:03:35 -08:00
09a906cff5
Bug 1322459 part 1 - Remove duplicate mRefCnt in CaptureSinkFilter. r=Ehsan
...
MozReview-Commit-ID: IdIjzeMI9fI
--HG--
extra : rebase_source : 340cc0fccac0e986cfb3949e78b55e0ee930a139
2016-12-13 15:48:18 +08:00
a7d2d90fd4
Bug 1315283: allow VP9 encoder in webrtc to reconfigure if the input resolution changes r=TD-Linux
2016-11-12 02:57:17 -05:00
e3634f0da1
Bug 1311048 - Surface process id in application sharing for completeness (we never share own process). r=jesup
...
MozReview-Commit-ID: HShFSBMO0yx
--HG--
extra : rebase_source : e1e3f2b967d72b37995d78a00aa9abb215e3eacf
2016-11-04 00:59:47 -04:00
0c777848be
Bug 1311048 - Make it compile on android. r=jesup
...
MozReview-Commit-ID: 7jwGgr0JVEt
--HG--
extra : rebase_source : 3c6e54e693d36a2236ff87304be02b388edecd87
2016-11-04 11:44:32 -04:00
f6022d3538
Bug 1311048 - Pick out Firefox windows as scary by pid on Linux. r=jesup
...
MozReview-Commit-ID: CjkfnZWjCTl
--HG--
extra : rebase_source : 54efcf2da32fc3a673903765b3e888d49a745c24
2016-11-04 23:56:48 -04:00
695cfd5856
Bug 1311048 - Pick out Firefox windows as scary by pid on Windows. r=jesup
...
MozReview-Commit-ID: DzlAVt1oJIs
--HG--
extra : rebase_source : 640dce53420d5b5d862244ef9d953cff899f1d1c
2016-11-03 17:21:56 -04:00
409780816c
Bug 1311048 - Pick out Firefox windows as scary by pid (only OSX atm). r=jesup
...
MozReview-Commit-ID: DwUodsRWswm
--HG--
extra : rebase_source : 0ab4002dfbbb4a0cf0265764d22c598256198d82
2016-09-30 13:23:42 -04:00
e2f31f9f1e
Bug 1273734 - [AVFoundation] Expose different FPS range as discrete capability; r=jib
...
MozReview-Commit-ID: 3k3r87VDhDR
--HG--
extra : rebase_source : 477e5e09ab7bc1b2b2685fc161f5336df9b7d553
2016-11-03 16:17:25 +08:00
0a7933f196
Bug 1310956 - Use arm64 for target_arch into GYP. r=glandium
...
Google's projects using GYP use arm64 for target_arch instead of aarch64. So we should use it for moz.build generator.
MozReview-Commit-ID: J4SLKhCqyUo
--HG--
extra : rebase_source : c902736ba0248eb5a3dfe94c174cb96374ebb94c
extra : histedit_source : 52620e73d6457078b28402dc6ef33f78f1c5425d
2016-10-21 14:48:24 +09:00
19b1cf1c88
Bug 1312431 - Make buffers passed into Get10msTone match specified maximum buffer size; r=jesup
...
MozReview-Commit-ID: I7UNWfzHi6I
--HG--
extra : rebase_source : b51572e3784685df3fd69f3560729e73c5fb9c1d
2016-10-28 08:58:19 -04:00
143157e3ea
Bug 1312431 - Support for higher sample rates in dtmf_inband.cc can not be reached; r=jesup
...
This also fixes a potential buffer overflow as the buffer size
was hard coded to be the old maximum buffer size of 320.
MozReview-Commit-ID: 5DD8vWlIDPQ
--HG--
extra : rebase_source : 60a4406510409f4080362061fd770c16ce534fa6
2016-10-24 11:36:29 -04:00
c1187fc9f8
Bug 1308412 - fix setCaptureHeight() failure; r=jesup,jib
...
MozReview-Commit-ID: 47mgoY4sOSw
--HG--
extra : rebase_source : f63ed9b3a39e3251accd7a6c35a6a4332db6c773
2016-10-26 18:08:51 +08:00
ed70e2a07e
Bug 1308412 - retrieve CVImageBuffer from CMSampleBuffer if it doesn't contain any CMBlockBuffer; r=jesup,jib
...
MozReview-Commit-ID: BERDnKJ0i3t
--HG--
extra : rebase_source : e5b41547e2f7ddf5f4bd1759e701176ce83e7d90
2016-10-26 18:08:18 +08:00
8cf1367dd8
Merge m-i to m-c, a=merge
...
MozReview-Commit-ID: FnnOWQ3xKPi
2016-10-25 22:03:31 -07:00
2837f878f5
Bug 1309469 - Crash [@ webrtc::ViECaptureImpl::NumberOfCaptureDevices]; r=jesup
...
MozReview-Commit-ID: Di4uXmJe6oz
--HG--
extra : rebase_source : 9dcdeffea1566864f030d14d907a339b657768c1
2016-10-24 18:19:16 +08:00
f99a704b53
Bug 1312814 - Remove SDK_INT>8. r=sebastian
...
MozReview-Commit-ID: 5LcClh9REEH
--HG--
extra : rebase_source : 9376494b708a511e68f6589886a74f872e3de84c
2016-10-25 23:32:47 +05:30
84a83b607a
Bug 1264596 - GeckoApp: Remove mCameraView and related code. r=gcp,sebastian
...
MozReview-Commit-ID: ACgp4iIAigw
--HG--
extra : rebase_source : e0c92b6d8e68d82b7db3c195e22a457dbc43d95d
2016-10-20 19:27:59 +05:30
c7407c854f
Bug 1308792 - clear DeviceInfoDSSingleton::GetInfo() in DeviceInfoDS dtor to prevent illegal access crash; r=jesup,jib
...
MozReview-Commit-ID: Fy6Qu6sONgr
2016-10-16 01:15:13 +08:00
2844380bd4
merge mozilla-inbound to mozilla-central a=merge
...
--HG--
rename : media/gmp-clearkey/0.1/ClearKeyCencParser.cpp => media/psshparser/PsshParser.cpp
rename : media/gmp-clearkey/0.1/ClearKeyCencParser.h => media/psshparser/PsshParser.h
rename : media/gmp-clearkey/0.1/gtest/TestClearKeyUtils.cpp => media/psshparser/gtest/TestPsshParser.cpp
rename : media/gmp-clearkey/0.1/gtest/moz.build => media/psshparser/gtest/moz.build
2016-10-12 12:01:48 +02:00
0bbb96335e
Bug 1308792 - protect the critical section accessing device_info_cs_; r=jesup
...
MozReview-Commit-ID: GGxtHUPAk6N
--HG--
extra : rebase_source : 335cb7962dbd5e8f2900ca7984c24e327924d9ec
2016-10-11 14:30:28 +08:00
d53aa7abf7
Bug 1307433 - WebRTC: implement RFC6051 ("Rapid Synchronisation of RTP Flows") r=pkerr
...
Actually just the first part of it (sending RTCP at the start of a flow),
not the RTCPFB message
2016-10-11 15:04:39 -04:00
fed17ce3e1
bug 1305506 - Remove some cruft from a webrtc gyp file. r=jesup
...
Things seem to build OK without this, and it's breaking some new code I added in gyp_reader.
MozReview-Commit-ID: 6ccaXZ0mRTj
--HG--
extra : rebase_source : c1e8acb39f863b3ff62492cf70e74748cb74e795
2016-08-16 10:09:03 -04:00
fd7b7476c2
Merge m-i to m-c, a=merge
...
MozReview-Commit-ID: 93ZdJbK1x05
2016-10-06 19:58:18 -07:00
a62e23429f
Bug 1291715 - Add support for 44100 and 48000 Hz sample rates to DtmfInband; r=jesup
...
The DtmfInband class does not support sample rates above 32000. This adds
support for 44100 and 48000.
The 'a' coefficients were calculated in python as:
int(round(32768*math.cos(2*math.pi*f/fs)))
The 'ym2' coefficients were calculated in python as:
int(round(16383*math.sin(2*math.pi*f/fs)))
where f was in: [697, 770, 852, 941, 1209, 1336, 1477, 1633] and fs was in:
[8000, 16000, 32000, 44100, 44800].
The calculated values were slightly off the existing values at 8000 Hz,
but agreed at 16000 and 32000 Hz.
MozReview-Commit-ID: GIzyUSyecR4
--HG--
extra : rebase_source : edbde6e8c8b6cfd1c44c808022849c688364745b
2016-09-14 16:07:46 -04:00
2f628de7ca
Bug 1307254: Implement ::Stop() for screen/window/app capture r=jib
2016-10-05 16:15:19 -04:00
b2bb341033
Bug 1300468 - implement mediaDevices.ondevicechange for Windows; r=jesup
...
MozReview-Commit-ID: IhqmXsqeuba
2016-10-06 02:31:26 +08:00
036d6b9816
Bug 1288904: Clean up RID header extension support r=jesup,drno
2016-09-23 21:55:08 -04:00
9bd39c7f55
Backed out changeset 07cb69423014 (bug 1288904) for failing cpp unit test jsep_session_unittest. r=backout
2016-09-24 10:23:53 +02:00
26e9eaaec7
Bug 1288904: Clean up RID header extension support r=jesup
2016-09-23 21:55:08 -04:00
47bcd9e1ed
Backed out changeset 34d05a052148 (bug 1300665) for failures in test_peerConnection_simulcastOffer.html
2017-02-09 20:57:58 -08:00
3023d161fc
Bug 1300665: Add abs-send-time and toffset header extension usage and negotiation r=bwc
...
MozReview-Commit-ID: 3h9C8XziNky
2017-02-09 20:56:29 -05:00
a268210415
Bug 1302059 - use VIDIOC_ENUM_FMT & VIDIOC_ENUM_FRAMESIZES to query supported format & resolution; r=jesup
...
MozReview-Commit-ID: AFEeb9yrIzb
--HG--
extra : rebase_source : c3b6b71b167a897cdd14e54f24b3c495d5151dd5
2016-09-21 17:25:00 +08:00
4f54bca31e
Bug 1304558 - Unbreak BSD build after bug 1297337. r=jesup
...
MozReview-Commit-ID: 3r2cjZlODDo
--HG--
extra : rebase_source : 942e02c6339b6c6ba753d117fadb1f558bb48b94
2016-09-22 00:04:44 +00:00
c86b6515ad
Bug 1297337 - implement mediaDevices.ondevicechange for Linux; r=jesup
...
MozReview-Commit-ID: 6cEq7xVUkhf
--HG--
extra : rebase_source : ee71fea0aa49452bcd403678b3c22c3fe3dd297c
2016-09-01 11:06:49 +08:00
6ec5958a84
bug 1302935: enable vp9 in webrtc and fix missing gof fields in codecSpecific r=pkerr,drno
...
Also required fixing tests to handle more codecs
2016-09-15 21:17:09 -04:00
f6ee0a3336
Backed out changeset 27f8a2467b31 (bug 1302935) for jsep_session_unittest failures
...
CLOSED TREE
2016-09-15 19:37:25 -07:00
5f4f5f0fc6
bug 1302935: enable vp9 in webrtc and fix missing gof fields in codecSpecific r=pkerr,drno
...
Also required fixing tests to handle more codecs
2016-09-15 21:17:09 -04:00
952bb47f7e
Backed out changeset cd0093211582 (bug 1302935) for mda bustage (bad tests assume VP8 only)
...
on a CLOSED TREE
2016-09-15 13:52:36 -04:00
c93d4f5ef3
bug 1302935: enable vp9 in webrtc and fix missing gof fields in codecSpecific r=pkerr
2016-09-15 09:33:37 -04:00
a974d905be
Bug 1297808: Limit combined size of RTCP APP packets r=drno
2016-08-29 13:23:51 -04:00
62d1bf1089
Merge m-c to a CLOSED TREE m-i
2016-08-23 22:57:10 -07:00
c1a667e1dc
Bug 1296798 - provide PYTHON value to run sub-commands with same Python; r=ted
...
MozReview-Commit-ID: 4fLNhPLk5fu
2016-08-23 14:58:25 -07:00
023745dee7
Bug 1296798 - upgrade gyp from upstream; r=ted
...
MozReview-Commit-ID: GeVBrUGbaFU
--HG--
rename : media/webrtc/trunk/tools/gyp/test/compiler-override/compiler.gyp => media/webrtc/trunk/tools/gyp/test/compiler-override/compiler-exe.gyp
rename : media/webrtc/trunk/tools/gyp/test/mac/app-bundle/TestApp/English.lproj/InfoPlist.strings => media/webrtc/trunk/tools/gyp/test/ios/app-bundle/TestApp/English.lproj/InfoPlist-error.strings
rename : media/webrtc/trunk/tools/gyp/test/mac/app-bundle/TestApp/English.lproj/InfoPlist.strings => media/webrtc/trunk/tools/gyp/test/mac/app-bundle/TestApp/English.lproj/InfoPlist-error.strings
rename : media/webrtc/trunk/tools/gyp/test/mac/gyptest-postbuild-static-library.gyp => media/webrtc/trunk/tools/gyp/test/mac/gyptest-postbuild-static-library.py
rename : media/webrtc/trunk/tools/gyp/test/rules/src/subdir4/asm-function.asm => media/webrtc/trunk/tools/gyp/test/rules/src/subdir4/asm-function.assem
2016-08-23 14:58:20 -07:00
9afd64fd90
Bug 1292500 - Notify WebrtcUI when video capturing is paused/resumed. r=gcp
...
MozReview-Commit-ID: UkJVR7zCbI
--HG--
extra : rebase_source : cf18e6d233d3cdda42d363d240f5375e79ef356d
2016-08-23 12:36:17 +02:00
01c4e8cc1b
Merge inbound to m-c. a=merge
2016-08-19 09:52:53 -04:00
a58f8b89a0
Merge m-c to inbound, a=merge
2016-08-18 16:32:58 -07:00
4eb2d3e90d
Bug 1286429 - implement mediaDevices.ondevicechange for Mac OSX; r=jib,smaug
...
MozReview-Commit-ID: D1Jr6I4qPyr
--HG--
extra : rebase_source : 0f4a97da80d25923c9b6f6550b94039aefa88de5
2016-08-12 01:04:49 +08:00
65347007cd
Bug 1295687: Cherrypick -Wunused-private-field warning fix from upstream gtest into our gtest clone for webrtc. r=jesup
...
MozReview-Commit-ID: Jz3baQ3smFY
--HG--
extra : rebase_source : f8b754afa6f0fc591a5e3377dc2895fc15f0747c
2016-08-16 11:53:10 -07:00
c6f55e903d
Bug 1294407: Clean up H264 STAP-A handling r=pkerr
2016-08-17 16:31:58 -04:00
e9097643d5
Merge inbound to central, a=merge
2016-08-12 13:44:29 -07:00
05fc0cdaee
Bug 1274340 - Make RED and ULPFEC payload type match sdp values; r=jesup
...
To be able to send and receive video with FEC enabled it appears we need to
have matching constant values here and in sdp/sipcc/ccsdp.h.
MozReview-Commit-ID: LZzAyMW9eEu
--HG--
extra : rebase_source : 1b0588b53c3906659711ab39d51533ae38db2568
2016-06-30 12:20:04 -04:00
3559474034
Bug 1293422 - Add PacketizeMode0 to RtpPacketizerH264; r=jesup
...
We were previously using PacketizeFuA which stripped the NAL header. Since the
fragment fit in a single packet it would then be sent without any header
causing difficulties on the receiving side. This adds a PacketizeMode0 which
leaves the header intact.
MozReview-Commit-ID: 91rbveSuXtT
--HG--
extra : rebase_source : 95092f5e3cbb31f9c4697ed4fd272cd458eb4e94
2016-08-09 15:59:48 -04:00
734482b5f4
Bug 1042631: Bustage fix for typo when resolving nits rs=bustage
2016-07-19 17:13:01 -04:00
1c7377b2ae
Bug 1042631: Fix Linux mouse position when sharing a window in WebRTC screensharing r=ng
2016-07-19 16:07:32 -04:00
71ac49183e
Bug 1285501 - Build linux/ directory on DragonFly, NetBSD and Solaris as well. r=jesup
...
MozReview-Commit-ID: 46Z55h9oWIm
--HG--
extra : transplant_source : %AE%E7%0D%9EHg%84%17Z%07%7D%12%95%C1A3%F5%ECw%A1
2016-07-08 09:27:08 +00:00
ae674679d8
Bug 1286644: cherry-pick AEC fix from Chromium Issue 576624 r=pkerr
2016-07-13 17:47:00 -04:00
3cf9863b59
Bug 1273652: Always reinitialize the receiver/jitterbuffer when reseting video decoding r=pkerr
2016-07-12 15:42:59 -04:00
b6bb031a33
Bug 1279036 - retrieve CMSampleBuffer via CMSampleBufferGetDataBuffer for mjpeg case. r=jib
...
--HG--
extra : rebase_source : 66513f41aefad8a03959dad10e59c64d62d48c71
2016-06-14 06:30:26 +01:00
8d9d9ee459
Bug 1210660 - Change vp8 threshold for static images in screensharing mode; r=jesup
...
This changes the static threshold in screensharing mode and ensures that the
screensharing mode is in fact passed to the codec.
This also causes the peer connection to update the media pipelines when a track is
replaced to cause the codec to be notified that the source has changed and to
change settings appropriately. It seems to be a common use case to have a camera
track be replaced by a screenshare track during a call.
MozReview-Commit-ID: HbV14uL4kIL
--HG--
extra : rebase_source : 34d9fff2efb777bdfd5887db879184bc4ffc7442
2016-06-09 13:38:43 -04:00
53c5fb9e43
Bug 1167544 - OpenH264 should not send STAP-A aggregation packets in mode 0; r=jesup
...
MozReview-Commit-ID: IY2mu9dVKHK
--HG--
extra : rebase_source : 77e455651ebf39cf56ca9892be59bd50622b8cc3
2016-06-01 14:28:43 -04:00
967e316e57
Bug 1278476 - Add android/aarch64 target to WebRTC's GYP. r=jesup
...
MozReview-Commit-ID: DzAZdAvsM9O
--HG--
extra : rebase_source : 1f4211a27382fb05abcae739558903da77fc57cb
2016-06-07 15:51:46 +09:00
2a354943cf
Bug 1277853 - adjust checks for clang's integrated assembler in transform_neon.S; r=jesup
...
This file used __APPLE__ as a shorthand for "compiling with clang's
integrated assembler"; we should really be checking for __clang__ instead.
2016-06-06 16:59:35 -04:00
21cc78c1a6
Bug 1276156: Stop encoder ProcessThreads before deleting the channel r=pkerr
...
To avoid deadlocks between DeleteChannel() and Process() code
MozReview-Commit-ID: G650SWDH8s0
2016-05-31 23:39:34 -04:00
1d9c7716f5
Bug 1275217: remove QuickTime and QTKit related code and dependecies. r=jib
...
MozReview-Commit-ID: IDXgV9jnlMk
--HG--
extra : rebase_source : 529d867eb7330b9f498897df7248221836a91016
2016-05-24 02:15:19 -07:00
f71af76062
Bug 1180725 - use AVFoundation for camera capture on OSX. r=jib
2016-05-19 22:48:55 +08:00
fa238a23db
Bug 1271041 - Switch DragonFly and NetBSD to pthread_condattr_setclock. r=jesup
2016-05-07 22:44:00 +02:00
7a9638bef9
Bug 1269165 - Restore ALSA plugins detection on non-Linux after bug 757637. r=jesup
...
--HG--
extra : rebase_source : b655bc3c2928c18ab8ed3fb170926ea8213a122b
2016-04-30 21:52:00 -04:00
2000e8ffcb
Bug 1269930: don't crash if an AEC logfile fails to open r=pkerr
...
MozReview-Commit-ID: 4MgOZe5jO3p
2016-05-04 11:48:18 -04:00
8ea71955dd
Bug 1269930: Fix some errant (though working) ifdefs in mods to upstream webrtc code r=pkerr
...
MozReview-Commit-ID: 3bCBD3I4fHO
2016-05-04 11:47:44 -04:00
6f22cfc9fb
Bug 1247574: Force webrtc audio input processing to resample to target rate to fix 16KHz mics. r=padenot
...
MozReview-Commit-ID: BBZcX03Z6Kn
2016-03-19 16:05:13 -04:00
d6e073cec4
Bug 1254876: assert windows recording is shut down r=pkerr
2016-03-21 02:57:13 -04:00
3101dc7152
Bug 820972 - Comment out colorTable[] because we don't need it. r=jesup.
...
This saves 64 KiB of static data.
2016-03-11 12:41:30 +11:00
568a64e79a
Bug 1254507 - Fix leak in WebRTC DesktopApplication class. r=jesup
...
MozReview-Commit-ID: FLuQZcPyv0d
--HG--
extra : rebase_source : 24bc9507bb4abee94a1238176675948249c102f1
2016-03-08 14:14:05 +01:00
9e95296e04
bug 1241064 - updating stats filter SSRC when audio channel SSRC changes; r=jib
...
bug: getStats was returning statistics for the shortlived, initial SSRC
now updating SSRC filter on statistics update callback to match audio channel ssrc
getStats API now returns statistics for correct SSRC: jitter, packets lost, etc.
MozReview-Commit-ID: WCd71WMkUW
--HG--
extra : rebase_source : 5d3a5a14e04313749173d264894e44411c3417bf
2016-03-03 08:03:06 -08:00
1ba1737ec0
Bug 1252699 - Set WEBRTC_DETECT_ARM_NEON when optional neon is requested. r=jesup
2016-03-03 06:28:10 +09:00
ceba79f2b0
Bug 1244913 - resolution-based bitrates for each simulcast layer, scaleResolutionDownBy, and working maxBitrate in unicast. r=bwc,jesup
...
MozReview-Commit-ID: 347J1ElsOEx
--HG--
extra : rebase_source : 33eff52e6082815d732de49a2bac584cfc9c87c4
2016-02-12 19:56:56 -05:00
831c3445c4
Bug 1248335: avoid using SvcInternal structure entirely, as system-vpx may not have it r=pkerr
...
MozReview-Commit-ID: 146FTSGQ8Ck
2016-02-23 11:55:24 -05:00
Dan Minor
Jean-Yves Avenard
Dan Horák
Michael Froman
Wes Kocher
Munro Mengjue Chiang
Randell Jesup
Nico Grunbaum
Chris Manchester
Mike Hommey
Tom Ritter
Eric Rahm
Paul Adenot
Makoto Kato
Heiher
Carsten "Tomcat" Book
Kartikaya Gupta
Nathan Froyd
Nicholas Nethercote
Jan-Ivar Bruaroey
Phil Ringnalda
Martin Stransky
Chris Peterson
Jan Beich
Sebastian Hengst
Masatoshi Kimura
Jim Chen
Nils Ohlmeier [:drno]
Florian Queze
John Paul Adrian Glaubitz
Ryan VanderMeulen
Ting-Yu Chou
Tushar Saini (:shatur)
Ted Mielczarek
Myk Melez
Sebastian Kaspari
Daniel Holbert
Nathan Froyd
Gian-Carlo Pascutto
[:ng]