fe5d36b6fe
Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well.
...
We will do some refactoring of video engine and would like to use the
same rtcp stats struct there. Both video and audio seem to use 8bit
fraction lost, so that is changed in the struct as well.
BUG=
R=henrik.lundin@webrtc.org , kjellander@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5039 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28 09:21:07 +00:00
c94abd313e
Use clang-format -style=chromium to correct the format in webrtc/modules/interface/module_common_types.h
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5036 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-25 18:15:09 +00:00
0729460acb
Added a "interleaved_" flag to webrtc::AudioFrame.
...
And also did some format refactoring on the AudioFrame class, no change on the functionalities on those format refactoring code.
BUG=
TEST=compile
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5032 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-25 12:50:46 +00:00
b56d0e383e
Change the low-bitrate handling in BitrateControllerImpl
...
Changing to using strategy classes rather than having two different
derived classes of BitrateControllerImpl. This enables run-time switching
of the strategy, which is now possible through a new API. The reason is
that it must fit the current design of ViE.
BUG=2436
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5028 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-24 09:24:06 +00:00
37bb4974e7
Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals.
...
R=juberti@google.com , mikhal@webrtc.org , stefan@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5027 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-23 23:59:45 +00:00
22858d4785
Add an extended filter option to audioproc.
...
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2609005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5024 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-23 14:07:17 +00:00
042e91c2b2
Fix for incorrect RTT estimation. A too low RTT value could be estimated.
...
R=andrew@webrtc.org , holmer@google.com , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2579005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5023 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-23 13:58:31 +00:00
31628aae7e
Upgrade scoped_ptr to Chromium's latest version.
...
Analogous to the recent libjingle change: http://cl/54929753-p10 .
This supports scoped_ptr<T[]> and scoped_ptr<C, FreeDeleter> rather
than scoped_array and scoped_ptr_malloc respectively.
- Add Chromium's template-based COMPILE_ASSERT. We didn't have this
previously in order to support the macro in C. Instead, move the
existing macro to compile_assert_c.h.
- Additionally copy the move.h and template_util.h depedencies and add
the WARN_UNUSED_RESULT macro.
- Leave scoped_array and scoped_ptr_malloc for now, but mark as
deprecated.
- Remove scoped_ptr foo(NULL) use. The default constructor handles it.
- Remove the now redundant COMPILE_ASSERT from peerconnection_jni.cc.
- Add a CHECK_ARRAY_SIZE macro to rtp_format_vp8_unittest.cc to remove
some repeated code.
TESTED=trybots
R=pbos@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2449005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5015 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-22 12:50:00 +00:00
621df678c8
WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN.
...
Mostly to remove a long-standing TODO...
TESTED=trybots
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5013 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-22 10:27:23 +00:00
943e3b95a6
Add CurrentLayerId() to temporal layers.
...
same patch as: https://webrtc-codereview.appspot.com/2427004/
TBR=holmer@google.com
Review URL: https://webrtc-codereview.appspot.com/2729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5012 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-22 01:55:07 +00:00
8215106371
Framework for testing bandwidth estimation.
...
BUG=
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2317004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5008 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-21 14:23:26 +00:00
29dd0de5b3
Changing the bitrate clamping in BitrateControllerImpl
...
This CL implements an alternative to the bitrate clamping that is done
in BitrateControllerImpl. The default behavior is unchanged, but if
the new algorithm is enabled the behavior is as follows:
When the new bitrate is lower than the sum of min bitrates, the
algorithm will give each observer up to its min bitrate, one
observer at a time, until the bitrate budget is depleted. Thus,
with this change, some observers may get less than their min bitrate,
or even zero.
Unit tests are implemented.
Also fixing two old lint warnings in the affected files.
This change is related to the auto-muter feature.
BUG=2436
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2439005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5007 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-21 14:00:01 +00:00
e05362916c
Make sure the first frame isn't dropped.
...
If frames were delivered within the same millisecond as VideoCaptureImpl
was created, or the timestamp weren't granular enough then the first
frame would be mistakenly dropped because of having the same timestamp
as a previous one, even though there was no previous one.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5004 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-21 09:02:30 +00:00
89b1e688ca
Minor comment fix after clang reformat.
...
TBR=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4996 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-18 14:23:29 +00:00
2df89c0c8b
MouseCursorMonitor implementation for OSX and Windows.
...
BUG=crbug.com/173265
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2388004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4994 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-17 19:47:18 +00:00
d030972139
Remove unused kPowTableFrac which causes anroid clang build failure.
...
http://build.chromium.org/p/tryserver.chromium/builders/android_clang_dbg/builds/84322/steps/compile/logs/stdio
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2417004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4981 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-16 20:32:09 +00:00
e6e749da38
Add MouseCursorRenderer.
...
The new class acts as a wrapper for DesktopCapturer interface. It takes
mouse shape and position from MouseCursorCapturer and renders it on the
frames produced by underlying DesktopCapturer.
BUG=crbug.com/173265
R=wez@chromium.org
TBR=andrew@webrtc.org (modules.gyp)
Review URL: https://webrtc-codereview.appspot.com/2387004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4968 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-16 02:48:41 +00:00
2767b53f66
Add MouseCursorCapturer interface with implementation for X11.
...
The new interface will be used to capture cursor shape and position and
blend it into the image captured with desktop capturers.
BUG=crbug.com/173265
R=wez@chromium.org
TBR=andrew@webrtc.org (modules.gyp)
Review URL: https://webrtc-codereview.appspot.com/2386005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4967 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-16 02:42:38 +00:00
3555303cb0
Roll chromium_revision 226126:228675 and fix clang warnings
...
By request from thakis@chromium.org , I disabled the
-Wno-unused-const-variable setting that is set in Chromium's
common.gypi so we can prepare our code for it's removal.
This required some cleanup in order to get the code to compile
with Clang having the -Wunused-const-variable warning enabled.
TEST=all trybots passing
BUG=none
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2400004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4966 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-15 20:10:17 +00:00
e5021fe590
Make RtpData and RtpFeedback destructors public.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4965 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-15 10:38:40 +00:00
c2e471d8b3
Compile out unused kMinTrustedDelayMs.
...
TBR=niklas.enbom@webrtc.org
TESTED=trybots
Review URL: https://webrtc-codereview.appspot.com/2398004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4963 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-15 02:11:21 +00:00
1871dd2fb7
NetEq4: Removing templatization for AudioVector
...
This is the last CL for removing templates in Audio(Multi)Vector.
BUG=1363
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2341004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4960 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-14 20:33:25 +00:00
30792987b8
Remove empty line in SharedXDisplay::RemoveEventHandler.
...
TBR=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2397004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4958 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-14 17:58:46 +00:00
05773e5a70
Android OpenSlDemo: remove some usages of deprecated APIs that is breaking the bots.
...
TBR=fischman@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/2395004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4956 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-14 16:25:11 +00:00
7419a72383
Add event handling in SharedXDisplay.
...
SharedXDisplay has to handle X events because the events may belong to
different clients of that class.
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2386004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4953 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-13 00:44:09 +00:00
894e6fe9ea
Add DesktopCaptureOptions class.
...
The new class is used to pass configuration parameters to screen/window
capturers. It also allows to share X Window connection between multiple
objects.
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2374004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4952 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-12 22:40:05 +00:00
f53622d42e
WebRTCDemo: Fixes warning for devices with pre-17 API level. Also fixes broken build build.xml and project.properties.
...
BUG=2083
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2375004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4951 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-11 21:28:26 +00:00
13b2d46593
clang-format audio_processing/aec/*
...
TBR=bjornv
TESTED=trybots
Review URL: https://webrtc-codereview.appspot.com/2373004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4944 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-08 23:41:42 +00:00
ca764ab22d
Add a parameter to audioproc for overriding the delay.
...
Rename the parameter for adding to the input delay to "add_delay".
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2345007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4939 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-07 16:44:32 +00:00
f5d7c5891c
Revert r4934 "Add a tool for parsing an RTP file and outputting the BWE relevant fields."
...
Revert r4935 "Fix build error in r4934."
TBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2364004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4936 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-07 08:42:46 +00:00
611e5141cb
Fix build error in r4934.
...
TBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2363004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4935 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-07 08:35:36 +00:00
bc99bcfa6f
Add a tool for parsing an RTP file and outputting the BWE relevant fields.
...
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2237005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4934 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-07 08:21:24 +00:00
6d5d248075
Wrap ACM2 code inside acm2 namespace. This gurantees that one ACM would not use components of others by accident.
...
BUG=
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2344004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4933 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-06 04:47:28 +00:00
f31639612d
Accounting for wrap-around of timestamps.
...
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2340006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4932 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-06 02:21:24 +00:00
35e4dd3067
VPM: Fixing namespace
...
R=marpan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2355004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4930 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-04 20:21:30 +00:00
4598380860
Android: enable camera video stabilization when available.
...
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2347005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4929 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-04 20:14:19 +00:00
3f9288f987
Add APK and isolate target for video_engine_tests
...
Add .isolate file and _run target for video_engine_tests.
Move tools/swarm_client to be untracked in all .isolate file,
so refactorings in swarm_client doesn't require us updating
all our .isolate files (similar to the changes for the
Chromium tests done in:
https://src.chromium.org/viewvc/chrome?view=rev&revision=218844 )
Update modules_unittests.isolate with new NetEq4 reference files
needed.
TEST=trybots passing
I also setup a Chromium workspace where I patched third_party/webrtc
with the changes in this CL, followed by compiling with the settings
described in
https://code.google.com/p/webrtc/issues/detail?id=1882#c11
I then verified that the video_engine_tests_apk dir was created
in the output folder.
BUG=1916,2462
R=andrew@webrtc.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2344007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4925 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-04 18:20:38 +00:00
acb00505b6
Only declare kDelayDiffOffset when used.
...
And remove the redundant Windows block.
R=hans@chromium.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2351004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4922 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-04 16:59:17 +00:00
ad2eb6f67d
Unbreaks Android build after r4915.
...
TBR=ajm@webrtc.org
BUG=Not filed
Review URL: https://webrtc-codereview.appspot.com/2348005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4921 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-04 14:21:23 +00:00
be9c560aab
Revert r4913 that reverts r4911. Original CL description:
...
"Adding temporal layer strategy that keeps base layer framerate at an acceptable value."
R=holmer@google.com , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2351006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4920 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-04 13:11:31 +00:00
6c82e04cee
Android standalone: remove some usages of deprecated APIs and prevent further regressions.
...
Also:
- Fixed WebRTCDemo UI to say "SwitchToBack" at startup since default camera is front
- Rebuild WebRTCDemo APK when resources/layout/strings change
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2337004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4916 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03 18:57:48 +00:00
4e65e07e41
VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android.
...
Besides being ~40% the size of the previous implementation, this makes it so
that VideoCaptureAndroid can stop and restart capture, which is necessary to
support onPause/onResume reasonably on Android.
BUG=1407
R=henrike@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2334004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4915 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03 18:23:13 +00:00
44db9d1a57
Revert 4911 "Adding temporal layer strategy that keeps base laye..."
...
> Adding temporal layer strategy that keeps base layer framerate at an acceptable value.
>
> R=stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/2272005
TBR=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4913 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03 17:42:07 +00:00
b43d8078a1
Reformatting VPM: First step - No functional changes.
...
R=marpan@google.com , marpan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2333004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4912 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03 16:42:41 +00:00
26f78f7ecb
Adding temporal layer strategy that keeps base layer framerate at an acceptable value.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2272005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4911 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03 14:06:14 +00:00
7ee3efb0d8
Disable Receiver unittests on Android.
...
BUG=
TBR=minyue@google.com
Review URL: https://webrtc-codereview.appspot.com/2344005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4909 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03 00:05:15 +00:00
6ea3d1cc9e
ACM test are modified to run with both ACM1 and ACM2.
...
Beside the changes in test files. acm2/acm_generic_codec.cc and acm2/audio_coding_module_impl.cc are modified to fix a bug.
Also, nack{.cc, .h, _unittest.cc} are removed form main/sourc as nack files in both ACM1 and ACM2 are essentially identical.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2192005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4908 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-02 21:44:33 +00:00
2a97317953
Fix include of isolate.gypi
...
Recent changes in GYP seem to have broken our previous
"hack" for getting the GYP rule for .isolate files
imported from the Chromium build/isolate.gypi.
The best solution for now is to remove the hack
and check in a copy of Chromium's src/build/isolate.gypi
in WebRTC's build/ dir instead. A similar approach is
used for our build/protoc.gypi file.
TEST=On Linux, I successfully ran:
gclient runhooks
ninja -C out/Release
and verified a bunch of .isolated files were created in
out/Release (which didn't happen before this patch).
I also renamed the build/isolate.gypi from Chromium to
ensure that our own is used and not that one (in case any
paths would be incorrect).
I also ran build/gyp_chromium in a Chromium checkout
with WebRTC in third_party/webrtc having this patch applied
to ensure GYP processing was still working.
Finally, I verified that the same project generation and
compilation from a Chromium checkout worked the way we build
our Android native tests, using:
. build/android/envsetup.sh
GYP_DEFINES="$GYP_DEFINES include_tests=1 enable_tracing=1" gclient runhooks
ninja -C out/Release android_builder_webrtc
BUG=1916
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2338004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4907 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-02 19:31:16 +00:00
f8f78b1316
Android OpenSL: Fixes faulty assertion in jni-code.
...
BUG=2452
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2342004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4906 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-02 18:41:06 +00:00
4887114af7
Remove templatization of the AudioVector test
...
This CL converts the unit tests for AudioVector from typed tests to
regular tests. It is in preparation for removing templatization for
AudioVector in an upcoming CL.
BUG=1363
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2319005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4903 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-02 15:07:28 +00:00
