guoweis@webrtc.org
c51fb9348d
Fix an assert failure caused by race condition
...
BUG=
R=pbos@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7938 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 00:30:55 +00:00
andrew@webrtc.org
0ab42bc3f6
Make safe_conversions suitable for rtc_base_approved.
...
Since we want to use checked_cast in WavReader.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7937 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 22:56:09 +00:00
pthatcher@webrtc.org
bc03192560
Move jingle examples from talk/ into webrtc/libjingle. This is part of the effor to move Jingle out of WebRTC and into its own repository.
...
R=juberti@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7936 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 22:15:11 +00:00
guoweis@webrtc.org
0eb6eec5cb
Move VirtualSocket into the .h file to allow unit tests more control over behavior.
...
BUG=3927
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7935 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 22:03:33 +00:00
aluebs@webrtc.org
6f10ae25ea
Support block_size greater than chunk_size in Blocker
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37399005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7934 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 17:28:31 +00:00
pbos@webrtc.org
eb544460e4
Rename _t struct types in audio_coding.
...
_t names are reserved in POSIX.
R=henrik.lundin@webrtc.org
BUG=162
Review URL: https://webrtc-codereview.appspot.com/34509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7933 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 15:23:29 +00:00
tommi@webrtc.org
209df9bf77
Change MockStatsObserver to grab values inside of OnComplete.
...
This is done since StatsReportCopyable is going away and the list of
supported properties of the mock class is known.
StatsReports holds a list of pointers to objects that cannot be cached,
so this is a simple way to grab the values when they're available.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7932 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 14:09:05 +00:00
pbos@webrtc.org
e728ee03ba
Remove or rename typedefs with _t prefixes.
...
_t prefixes are reserved for additional typenames in POSIX.
R=henrik.lundin@webrtc.org , hta@webrtc.org , stefan@webrtc.org
BUG=162
Review URL: https://webrtc-codereview.appspot.com/36559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7931 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 13:43:55 +00:00
tommi@webrtc.org
5263c58923
Add a little utility to capture cpu graphs.
...
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7930 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 12:35:29 +00:00
sprang@webrtc.org
70f74f3f7b
Add overshoot of target bitrate for screenshare with temporal layers.
...
Set the codec target bitrate higher than TL0 but lower than TL1, making
sure frame rate is not too low (but still lower than TL1) and that
overshooting for complex scenes don't overly exceed TL1 bitrates.
BUG=4083
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7929 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 10:57:10 +00:00
asapersson@webrtc.org
45a272ab22
Change aggregated fraction loss to be calculated from the cumulative loss and extended sequence number diff between the current and the last report block of two get stats calls.
...
Previously it was derived from the fraction loss of the current report (which could be based on a received report block in between two get stats calls).
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7928 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 10:27:57 +00:00
kwiberg@webrtc.org
e102e8147b
Enable the iSACfix AudioDecoder test (and make it work again)
...
As far as I can tell, the test should have been enabled again once
https://code.google.com/p/webrtc/issues/detail?id=1353 was fixed, but
it wasn't, and has rotted a bit as a result. I'm not sure why the
number of encoded bytes have changed, but the output seems to be
correct (EncodeDecodeTest encodes, decodes, and compares the result
with the original).
The DecodePlc change is necessary because r7912 added support for that
to the iSACfix AudioDecoder.
BUG=1353, 3926
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7927 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 07:30:23 +00:00
braveyao@webrtc.org
38881be912
If one of the bundled content is missing in SDP, return false to MaybeEnalbeMuxingSupport().
...
Verified in chromium. Now the existing content still could work.
BUG=4096
TEST=Manual Test
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7926 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 05:59:41 +00:00
guoweis@webrtc.org
950c518251
Add adapter_type into Candidate object.
...
Expose adapter_type from Candidate such that we could add jmidata on top of this.
Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.
This is migrated from issue 32599004
BUG=
R=juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7885
Committed: https://code.google.com/p/webrtc/source/detail?r=7906
Review URL: https://webrtc-codereview.appspot.com/36379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7925 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 23:01:31 +00:00
andrew@webrtc.org
971bf557e2
Fix path to mock_agc.h
...
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7924 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 22:28:20 +00:00
pthatcher@webrtc.org
f050791ba0
Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."
...
This reverts r7992.
It broke the Chromium build because the Chroumium build relies on the logic in webtc/libjingle/session.cc, but Chromium doesn't compile that file.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7923 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 22:28:03 +00:00
pthatcher@webrtc.org
4afb59903c
Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
...
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7922 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 21:37:37 +00:00
pthatcher@webrtc.org
e2b7585bc2
Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository.
...
R=juberti@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7921 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 21:09:08 +00:00
henrik.lundin@webrtc.org
a32487f97b
Disable AudioEncoderCopyRedDeathTest.NullSpeechEncoder
...
Fails linux memcheck.
BUG=4108
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7920 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 21:04:55 +00:00
pthatcher@webrtc.org
02c21dbef1
Make one OWNERS files for all of webrtc/libjingle so we don't need approval from webrtc/OWNERS every time we want to add a directory.
...
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7919 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 21:04:41 +00:00
andrew@webrtc.org
08df9b2841
Add a manageable command-line tool for AudioProcessing.
...
This is the start of a replacement for the venerable and unwieldly
process_test.cc (aka audioproc). It will be limited to:
- Reading WAV or aecdebug protobuf files.
- Calling the float AudioProcessing interface.
- Requiring aecdebug files for running bi-directional stream
components (e.g. AEC).
This initial version only handles WAV files.
R=aluebs@webrtc.org , bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7918 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 20:57:15 +00:00
aluebs@webrtc.org
cf6d0b64ef
Add 48kHz support to AGC
...
Doing the same for the 16-24kHz band than was done in the 8-16kHz.
Results look and sound as nice.
Originally reviewed here:
https://webrtc-codereview.appspot.com/26339004/
BUG=webrtc:3146
R=andrew@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7917 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 20:56:09 +00:00
andrew@webrtc.org
2510d11c0f
Add (safe) uint32_t cast to fix Win64 build.
...
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7916 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 20:47:42 +00:00
andrew@webrtc.org
048c5029f5
Handle all permissible PCM fields with WavReader.
...
I discovered the hard way that Adobe Audition writes an 18 byte format
header with an extra (zero) extension size field. Although:
https://ccrma.stanford.edu/courses/422/projects/WaveFormat/
indicates this field shouldn't exist for PCM, the documentation here:
http://www-mmsp.ece.mcgill.ca/documents/AudioFormats/WAVE/WAVE.html
doesn't list it as strictly forbidden, only that it _must_ exist for
non-PCM formats.
Audition can write metadata to the file after the audio data, which is
also not forbidden. We now ensure to read only up to the audio payload
length to avoid reading the metadata.
R=aluebs@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7915 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 20:17:21 +00:00
pbos@webrtc.org
451a133f44
Add AGC manager tests.
...
R=bjornv@webrtc.org
BUG=4098
Review URL: https://webrtc-codereview.appspot.com/35539005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7914 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 14:48:47 +00:00
henrik.lundin@webrtc.org
c1c9291e9b
Make an AudioEncoder subclass for RED
...
This class only supports the simple case of payload duplication. That
is, one single encoder is used, and the redundant payload is a one-step
delayed payload.
BUG=3926
R=kjellander@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 13:41:36 +00:00
kwiberg@webrtc.org
88bdec8c3a
AudioEncoder subclass for iSACfix
...
This patch refactors AudioEncoderDecoderIsac so that it can share
almost all code with the very similar AudioEncoderDecoderIsacFix.
BUG=3926
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7912 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 12:49:37 +00:00
kjellander@webrtc.org
0198933b3d
Cleanup: Remove 'const' qualifier from OnReceivedEstimatedBitrate().
...
This should fix the following error I'm seeing in Win8 GN trybot:
e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\bitrate_controller\bitrate_controller_impl.cc(78)
: error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\bitrate_controller\bitrate_controller_impl.cc(30)
: warning C4373:
'webrtc::BitrateControllerImpl::RtcpBandwidthObserverImpl::OnReceivedEstimatedBitrate':
virtual function overrides 'webrtc::RtcpBandwidthObserver::OnReceivedEstimatedBitrate',
previous versions of the compiler did not override when parameters only differed by const/volatile qualifiers
e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\rtp_rtcp\interface\rtp_rtcp_defines.h(286)
: see declaration of 'webrtc::RtcpBandwidthObserver::OnReceivedEstimatedBitrate'
http://build.chromium.org/p/tryserver.chromium.win/builders/win8_chromium_gn_dbg/builds/23/steps/compile/logs/stdio
The above was triggered in CL https://codereview.chromium.org/802113002/
BUG=None
R=kjellander@google.com , kjellander@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37409004
Patch from Thiago Farina <tfarina@chromium.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7911 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 12:29:59 +00:00
asapersson@webrtc.org
d08d389ce8
Add field to counters for when first rtp/rtcp packet is sent/received.
...
Use this time for histogram statistics (send/receive bitrates, sent/received rtcp fir/nack packets/min).
BUG=crbug/419657
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7910 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 12:03:11 +00:00
bjornv@webrtc.org
b395a5ea65
audio_processing: Moved legacy AGC code to webrtc/modules/audio_processing/agc/legacy/
...
include/ is renamed to legacy/ and analog_agc.* and digital_agc.* moved into the directory.
BUG=
R=andrew@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7909 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 10:38:10 +00:00
guoweis@webrtc.org
55360ae402
Revert "Add adapter_type into Candidate object."
...
This reverts commit aaf02cc2d4
.
BUG=
TBR=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7908 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 05:28:10 +00:00
marpan@webrtc.org
d021bbbc9e
Fix vp9 setting in vie loopback test.
...
If vp9 codec was selected then videoCodec.codecSpecific.VP8.numberOfTemporalLayers was being set.
TBR=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/37389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7907 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 00:21:47 +00:00
guoweis@webrtc.org
aaf02cc2d4
Add adapter_type into Candidate object.
...
Expose adapter_type from Candidate such that we could add jmidata on top of this.
Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.
This is migrated from issue 32599004
BUG=
R=juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7885
Review URL: https://webrtc-codereview.appspot.com/36379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7906 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 23:03:10 +00:00
pkasting@chromium.org
0b1534c52e
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
...
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.
This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".
BUG=chromium:81439
TEST=none
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 22:09:40 +00:00
aluebs@webrtc.org
96a626262a
Remove 20ms support in AGC
...
Today, 10ms is the standard chunk length used in whole AudioProcessing, so this was only adding unnecessary complexity and maintainance.
Removing it doesn't change the bahavior in any use case of today.
R=andrew@webrtc.org , bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7904 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 21:54:50 +00:00
guoweis@webrtc.org
1f05c45976
Reenable test case P2PTransportChannelTest.TestIPv6Connections
...
BUG=3317
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7903 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 21:25:54 +00:00
pbos@webrtc.org
a7f77720cb
Merge in AGC manager and AGC tools.
...
R=bjornv@webrtc.org
BUG=4098
Review URL: https://webrtc-codereview.appspot.com/37379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7902 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 16:33:16 +00:00
bjornv@webrtc.org
903b4ae603
Removes unused test files by audio_processing/transient
...
BUG=
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7901 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 16:13:05 +00:00
bjornv@webrtc.org
dd322136fe
resources/audio_processing: Removed unused test files
...
Two files not used by any tests are removed.
BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7900 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 15:57:11 +00:00
minyue@webrtc.org
6fd9308420
Suppressing warnings in GetRTT() in VoE.
...
GetRTT() was separated from GetRTPStatistics() but the warnings were not updated.
Now GetRTT() is only only used by GetRTPStatistics() and the warning pops up pointlessly and too often.
This CL is to suppress these warnings and maintain a proper warning for GetRTPStatistics().
BUG=
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7899 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 14:56:44 +00:00
tommi@webrtc.org
e2e199b894
Clean up StatsObserver's OnComplete methods (address TODOs).
...
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7898 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 13:22:54 +00:00
pbos@webrtc.org
3440fe1bc5
Use webrtc_root instead of DEPTH for iSAC.
...
Un-breaks chromium.webrtc.fyi. Broken as Chromium doesn't have webrtc/
checked out in root.
TBR=bjornv@webrtc.org ,tommi@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/28289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7897 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 10:56:50 +00:00
buildbot@webrtc.org
032b802a8c
(Auto)update libjingle 82121498-> 82126219
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7896 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:48:07 +00:00
tommi@webrtc.org
dd0601fbcf
Remove unneeded ctor and add a more practical one
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The default constructor isn't necessary, so I'm removing it.
I'm adding another one so that we can (later) make |type| const.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7895 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:47:49 +00:00
tommi@webrtc.org
69bc5a300f
Add thread asserts to StatsCollector.
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Also adding a "ForTest" postfix to a test-only method.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7894 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:44:48 +00:00
pbos@webrtc.org
788acd17ad
Merge audio_processing changes.
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R=aluebs@webrtc.org , bjornv@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/32769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7893 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:41:24 +00:00
pbos@webrtc.org
fb108b5a28
Revert r7885.
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Breaks compile step of other code where network name of
cricket::Candidate is used.
TBR=guoweis@webrtc.org ,juberti@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/31229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7892 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 08:04:50 +00:00
andrew@webrtc.org
b413a30097
Add WebRtcIsacfix_FilterMaLoopNeon's intrinsics version.
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This intrinsics version gives bit-exact result as the current assembly
neon code. And the performance is 38% better than current assembly
neon version, 5.92 times faster than current C version. The test runs
under Cortex-a53 aarch32 mode, other cpu should give similar performance
result.
BUG=4002
R=andrew@webrtc.org , jridges@masque.com
Change-Id: I257e33ef6d634a519fd71adc4f52b06dd655bd9d
Review URL: https://webrtc-codereview.appspot.com/32749004
Patch from Zhongwei Yao <zhongwei.yao@arm.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7891 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 07:23:49 +00:00
pbos@webrtc.org
18a3896bd2
Revert r7886:7887.
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Broke build steps in other code that uses securetunnelsessionclient.cc
and others.
TBR=tommi@webrtc.org ,pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/36439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7890 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 07:03:04 +00:00
andrew@webrtc.org
40e4767f2b
Add NEON intrinsics version for min_max_operations_neon.c
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WebRtcSpl_MinValueW32Neon, WebRtcSpl_MaxValueW32Neon, WebRtcSpl_MaxValueW16Neon
and WebRtcSpl_MaxAbsValueW32Neon are added. SplTest in common_audio_unittests
is passed on ARM32/ARM64 platforms.
BUG=4002
R=andrew@webrtc.org , jridges@masque.com
Change-Id: Id461d64c3313f56147edadd2231e8845574ead2a
Review URL: https://webrtc-codereview.appspot.com/28259004
Patch from Yang Zhang <yang.zhang@arm.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7889 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 06:07:47 +00:00