1f67b53c88
Remove dual stream functionality in ACM
...
This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. With this change, there is no longer need for the
ProcessDualStream method, which is removed. Consequently, the method
ProcessSingleStream is renamed to Process.
BUG=3520
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8074 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 09:36:30 +00:00
9ce01e6416
Clean unnecessary workaround for chromium import.
...
BUG=4185
R=kjellander@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8073 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 09:12:45 +00:00
0800db74b9
Add percentage of fec packets and recovered media packets to histogram stats:
...
- "WebRTC.Video.ReceivedFecPacketsInPercent"
- "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec"
BUG=crbug/419657
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8072 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 07:40:20 +00:00
61c1247224
Fix a case where empty candidate id is used
...
BUG=4161
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8071 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 06:53:07 +00:00
6c3855258d
Add WebRtcIsacfix_AllpassFilter2FixDec16Neon()'s intrinsics version.
...
This intrinsics version gives bit-exact result as the current C
code. And the performance is 14% better than current assembly
neon version, 3.4 times faster than current C version. The test runs
under Cortex-a53 aarch32 mode, other cpu should give similar performance
result.
Change-Id: Icce5eaf2e17790ce44513d52b53b9f600cc16f96
BUG=4002
R=andrew@webrtc.org , jridges@masque.com
Review URL: https://webrtc-codereview.appspot.com/36689004
Patch from Zhongwei Yao <zhongwei.yao@arm.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8070 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 02:56:06 +00:00
5a92b78e86
Add beamforming to audioproc_float utility.
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8069 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 01:28:36 +00:00
6b6301588e
Move ring_buffer to common_audio.
...
In preparation for adding a C++ wrapper in common_audio. Also, change
the return type of Init to void and call it from Create.
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8068 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 00:09:53 +00:00
fd630a50d2
Add BundlePolicy to RTCConfiguration. Don't change any behavior. Just make it possible to make progress in Chromium while we work on the behavior.
...
R=decurtis@webrtc.org , juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8067 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 23:19:06 +00:00
693e01c910
Fix searching for DirectX SDK during GN build.
...
Before that GN just checked for DXSDK_DIR environment variable.
GYP does more and checks registry, let's do the same in GN.
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37599004
Patch from Vyacheslav Chigrin <vchigrin@yandex-team.ru >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8066 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 21:25:25 +00:00
f1c8b90520
Remove WebRtcVideoEncoderFactory2.
...
This interface is no longer required and just adds complexity.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/33009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8065 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 17:29:27 +00:00
e5a31e1bf5
Revert removing of compile_assert.h.
...
In https://webrtc-codereview.appspot.com/39469004 compile_assert.h is removed and that resulted in some bots to break. There is a pending CL to fix the issue https://chromereviews.googleplex.com/141837013/
, meanwhile I revert this change.
TBR=kwiberg@google.com
Review URL: https://webrtc-codereview.appspot.com/35779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8064 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 17:17:11 +00:00
85fa94dff5
Exclude EndToEndTest.SendsAndReceivesH264 for Dr Memory.
...
This test is too slow to execute:
[ RUN ] EndToEndTest.SendsAndReceivesH264
e:\b\build\slave\win-drmem\build\src\webrtc\video\end_to_end_tests.cc(287): error: Value of: Wait()
Actual: 3
Expected: kEventSignaled
Which is: 1
Timed out while waiting for enough frames to be decoded.
[ FAILED ] EndToEndTest.SendsAndReceivesH264 (72812 ms)
BUG=3159
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8063 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 17:00:15 +00:00
387841ac5c
Improved fairness simulation by starting the flows 20 seconds apart.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8062 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 16:45:29 +00:00
f18fba2f7b
Implement SimulcastEncoderAdapter support.
...
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/37589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8061 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 16:26:23 +00:00
8315d7de85
Remove dual stream functionality in VoiceEngine
...
This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. The corresponding code in ACM will be deleted in a
follow-up CL.
BUG=3520
R=henrika@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8060 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 16:07:26 +00:00
b4e5d1b34e
Remove RTX SSRC when deleting the default receive stream.
...
BUG=crbug 448632
TEST=New unittest hitting assert without this change.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8059 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 15:07:07 +00:00
2ebfac5649
Remove COMPILE_ASSERT and use static_assert everywhere
...
COMPILE_ASSERT is no longer needed now that we have C++11's
static_assert.
R=aluebs@webrtc.org , andrew@webrtc.org , hellner@chromium.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8058 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 10:51:54 +00:00
86e1e487e7
Move system_wrappers.gyp files to the proper directory.
...
Build targets should not refer to non-subpaths as was happening before when
source/system_wrappers.gyp refers to ../interface/ files.
R=kjellander@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8057 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 09:30:52 +00:00
a35f741bb0
Add .classpath + talk/app/webrtc/androidtests to .gitignore
...
TBR=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8056 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 09:05:38 +00:00
f7a5893f80
Combine RegKeyTests to prevent parallel execution.
...
Executing these tests in parallel causes failures due to conflicting
registry keys, combining them to unblock launching a parallel win32 bot.
Ideally these keys would be generated differently per-process and not
conflict at all (so it can be run in parallel repeatedly alongside itself).
BUG=4162
R=kjellander@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8055 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 09:03:16 +00:00
ef090927f4
No longer asserting in mocks, split first test case in two methods.
...
This way assertions will be caught in the test runner instead of crashing other Android threads.
BUG=None
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8054 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 08:56:06 +00:00
69f47381fb
Roll chromium_revision 3dd2edf..a6eafec (310717:311223)
...
Relevant changes:
* src/third_party/android_tools: 8fe116f..56b3d3e
* src/third_party/boringssl/src: aac2f6a..ca9a538
* src/third_party/icu: 51c1a4c..4e3266f
* src/third_party/libvpx: d3f3dce..4f9bd1b
Details: 3dd2edf..a6eafecTBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41389005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8053 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 06:06:04 +00:00
d6e84d9d13
Always copy processed audio to output buffer in ProcessStream.
...
In the old AudioFrame ProcessStream API, input and output buffers were shared.
Now that the buffers are distinct, the input must be copied to the
output even when no processing occurred.
R=andrew@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=78de5010d167d1e375e05d26177aad43c2e2de08
Review URL: https://webrtc-codereview.appspot.com/41459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8052 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 01:33:54 +00:00
c0da63c707
Optimize minimum delay in blocker
...
Could not hear any difference when running the beamformer_test, although sample-wise it changes because of the non-linear character of the processing.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8051 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 22:28:35 +00:00
af9d56f38c
Unify the two copies of template_util.h
...
This patch basically deletes webrtc/base/template_util.h (which is the
more outdated copy, although there are only cosmetical differences)
and moves webrtc/system_wrappers/source/template_util.h to take its
place.
The reunified header uses the rtc namespace like the old
webrtc/base/template_util.h, rather than the webrtc namespace like
webrtc/system_wrappers/source/template_util.h.
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8050 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 20:32:04 +00:00
0b0c24177b
Only return Rtx mode in RTXSendStatus().
...
There is no need to return 'ssrc' and 'payloadtype' inside this function
since they are never used.
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38569004
Patch from Changbin Shao <changbin.shao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8049 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 14:15:15 +00:00
3df38b442f
Unify the two copies of compile_assert.h
...
This patch basically deletes webrtc/base/compile_assert.h (which is
the more outdated copy) and moves
webrtc/system_wrappers/source/compile_assert.h to take its place.
R=aluebs@webrtc.org , andrew@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8048 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 11:37:48 +00:00
58a1ba6ffc
Roll chromium_revision 271c6cc..3dd2edf (309333:310717)
...
Relevant changes:
* src/buildtools: 23a4e2f..451dcd0
* src/third_party/boringssl/src: 306e520..aac2f6a
* src/third_party/openmax_dl: 0164270..1a4171c
* src/tools/gyp: fe00999..82b0804
* src/tools/swarming_client: 119b084..c44f572
Details: 271c6cc..3dd2edfR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8047 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 10:59:21 +00:00
46323b3786
Remove useless AudioProcessing::Create() overload.
...
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8046 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 06:48:06 +00:00
16825b1a82
Use int64_t more consistently for times, in particular for RTT values.
...
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t. Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org , holmer@google.com , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 21:51:21 +00:00
a7add19cf4
audio_processing: Replaced macro WEBRTC_SPL_MUL_16_16 with * in high_pass_filter
...
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
BUG=3348,3353
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8044 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 21:12:29 +00:00
2a26734f04
Partial revert of r7396
...
This change reverts a small part of what was done in r7396. It seems
like that change uncovered another issue with NEON.
BUG=4177,chrome-os-partner:31534
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8043 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 20:52:21 +00:00
be40eb0579
Allow 720x1280 frames encoding on Android.
...
Current maximum encoder width and height for Android is
hard-coded to 1280x720, so if device is rotated to portrait
orientation only part 720x1280 camera frame is extracted and
scaled to 1280x720. Increasing maximum height to 1280 allows
feeding video encoder with rotated 720x1280 frames directly
without scaling.
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8042 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 17:55:47 +00:00
a525c98ca5
Fix parallelizability in ApmTests.
...
Using temporary filenames instead of fixed ones permits them to run in
parallel.
BUG=chromium:445880
R=andrew@webrtc.org , kjellander@webrtc.org
TEST=third_party/gtest-parallel/gtest-parallel -r100 -w100 out-asan/out/Debug/modules_unittests --gtest_filter=*ApmTest*:*CommonFormats*
Review URL: https://webrtc-codereview.appspot.com/35709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8041 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 17:31:18 +00:00
45db7eefa2
Use Java based audio as default for WebRTC.
...
The work landed in 4034 (use of HW AEC in AppRTC) is currently not
active by default since we build for Open SL. I missed that when I
did my initial change (since I always disabled OpenSL by GYP_DEFINES).
This CL ensures that Java based audio is used as default in WebRTC.
It would be great if we could shift over to Open SL (to cut latency)
but that would (today) mean that we can't support the HW AEC.
Hence, we are not ready to do so yet.
BUG=4034
R=kjellander@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8040 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 14:27:23 +00:00
81134d019d
Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory.
...
In order to do that, the signaling thread is also changed to wrap the current thread unless an external signaling thread thread is specified in the call to CreatePeerConnectionFactory.
This cleans up the PeerConnectionFactory and makes sure a user of the API will always access the factory on the signaling thread.
Note that both Chrome and the Android implementation use an external signaling thread.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8039 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 08:30:16 +00:00
88a4298234
common_audio: Made input vector const in WebRtcSpl_LevinsonDurbin()
...
In addition, expanded the unit test to verify both unstable and stable filters.
BUG=3353, 1132
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8038 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 05:53:43 +00:00
c14e3572c6
common_audio: Made input signal const in WebRtcSplFilterMAFastQ12()
...
BUG=3353, 1133
TESTED=locally on Mac and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8037 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 05:50:52 +00:00
19e4e8d751
Add support for trying alternate server (STUN 300 error message) on TCP
...
BUG=3774
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8036 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-10 02:41:32 +00:00
0ba1533fdb
Added support for an Origin header in STUN messages.
...
For WebRTC there are instances where it may be desirable to provide
information to the STUN/TURN server about the website that initiated
a peer connection. This modification allows an origin string to be
included in the MediaConstraints object provided by the browser, which
is then passed as a STUN header in communications with the server.
A separate change will be submitted to the Chromium project that
uses and is dependent on this change, implementing IETF draft
http://tools.ietf.org/html/draft-johnston-tram-stun-origin-02
Originally a patch from skobalt@gmail.com .
(https://webrtc-codereview.appspot.com/12839005/edit )
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8035 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-10 00:47:02 +00:00
2693a54614
Add WEBRTC_BEAMFORMER define to BUILD.gn
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8034 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 23:26:13 +00:00
8f27fcce79
Revert 8028 "Support associated payload type when registering Rt..."
...
Reasons for revert:
1. glaznev discovered potentially related problems using the Android AppRTCDemo.
2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky.
> Support associated payload type when registering Rtx payload type.
>
> Major changes include,
> - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
> - Receiver: Restore RTP packets by the new RTX-APT map.
> - Sender: Send RTP packets by checking RTX-APT map.
> - Add RTX payload type for RED in the default codec list.
>
> BUG=4024
> R=pbos@webrtc.org , stefan@webrtc.org
> TBR=mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/26259004
>
> Patch from Changbin Shao <changbin.shao@intel.com >.
TBR=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 20:22:46 +00:00
80452d70cb
Sync Android AppRTCDemo with internal repo.
...
- Fixed some Lint warnings.
- Switch to OPUS by default.
- Add check to WebSocket connection that public methods are called
on correct thread.
R=jiayl@webrtc.org , wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8032 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 19:34:06 +00:00
9657265f39
Revert "Accept incoming pings before remote answer is set to reduce connection latency."
...
This reverts r7980.
It was causing the ICE connected state to happen while still in the new state rather than going through the checking state, which was causing an ASSERT to fire, which was causing a crash.
Review URL: https://webrtc-codereview.appspot.com/41429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8031 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 19:08:27 +00:00
f3fd8e7cdf
Add NEON intrinsics version for transform_neon
...
WebRtcIsacfix_Time2SpecNeon and WebRtcIsacfix_Spec2TimeNeon are added.
TransformTest in modules_unittests is passed on ARM32/ARM64 platform.
Initially reviewed here:
https://webrtc-codereview.appspot.com/36449004/
BUG=4002
R=andrew@webrtc.org , jridges@masque.com
Change-Id: I0920ff66a0a0f529707fd7e6619f91e271a47019
Review URL: https://webrtc-codereview.appspot.com/31309004
Patch from Yang Zhang <yang.zhang@arm.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8030 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 18:29:37 +00:00
1592df78ef
PRESUBMIT: Add GN trybots for Windows and Mac.
...
Add to the default set since they're now green.
BUG=4105
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8029 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 15:38:29 +00:00
2a169640a3
Support associated payload type when registering Rtx payload type.
...
Major changes include,
- Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
- Receiver: Restore RTP packets by the new RTX-APT map.
- Sender: Send RTP packets by checking RTX-APT map.
- Add RTX payload type for RED in the default codec list.
BUG=4024
R=pbos@webrtc.org , stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26259004
Patch from Changbin Shao <changbin.shao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8028 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 15:16:10 +00:00
8649fed1b8
GN: Fix Windows build.
...
This required a tiny include fix in
src/third_party/winsdk_samples/src
which was committed in
https://code.google.com/p/webrtc/source/detail?r=7951
This incorporates contribution from vchigrin@yandex-team.ru
in https://webrtc-codereview.appspot.com/29299004/
BUG=261,1348,4105
R=pbos@webrtc.org
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8027 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-08 21:22:01 +00:00
2ead571fb6
Hard define the GUID for AudioEndpoint to avoid conflicts during compile.
...
BUG=3996
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8026 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-08 19:18:01 +00:00
758d6d431e
audio_processing/aecm: Removed usage of macro WEBRTC_SPL_MUL_16_16
...
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
BUG=3348, 3353
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8025 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-08 17:52:56 +00:00
