90a1cb4630
Revert 8810 "- Add a SetPriority method to ThreadWrapper"
...
Seeing if this is causing roll issues.
> - Add a SetPriority method to ThreadWrapper
> - Remove 'priority' from CreateThread and related member variables from implementations
> - Make supplying a name for threads, non-optional
>
> BUG=
> R=magjed@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/44729004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48609004
Cr-Commit-Position: refs/heads/master@{#8818}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8818 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-22 14:34:46 +00:00
346a64b9b5
Mac would force bluetooth playout working with 8kHz/1ch if capturing/rendering shares the same device, e.g. changing from 44.1kHz/2ch as default.
...
So in the HandleStreamFormatChange() callback, we need to re-initiate the playout as same as what we do in InitPlayout(). Here we merely copy those codes out from InitPlayout() into a new SetDesiredPlayoutFormat() function for the invoking from the two places.
Previously, HandleStreamFormatChange only re-creates the AudioConverter, which is not enough. We also need to reset the buffer size and refresh the latency.
BUG=4240
TEST=Manual Test
R=andrew@webrtc.org , henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36029004
Cr-Commit-Position: refs/heads/master@{#8815}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8815 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-21 01:06:14 +00:00
4553941d32
Document the 'int' return value of Resampler methods.
...
Remove an obsolete TODO comment.
R=andrew@webrtc.org
BUG=none
Review URL: https://webrtc-codereview.appspot.com/48589004
Cr-Commit-Position: refs/heads/master@{#8814}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8814 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 23:28:39 +00:00
3200a64b3c
Minor fix for MIPS Android build.
...
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47729004
Patch from Ljubomir Papuga <lpapuga@mips.com >.
Cr-Commit-Position: refs/heads/master@{#8813}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8813 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 22:55:43 +00:00
b9557a9bb7
Fix code to handle crashes for non-VP8.
...
Unit tests will be submitted Monday, submitting this part to get the
Android bots green.
BUG=1667, 1788
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44789004
Cr-Commit-Position: refs/heads/master@{#8811}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8811 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 19:53:15 +00:00
b6817d793f
- Add a SetPriority method to ThreadWrapper
...
- Remove 'priority' from CreateThread and related member variables from implementations
- Make supplying a name for threads, non-optional
BUG=
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44729004
Cr-Commit-Position: refs/heads/master@{#8810}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8810 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 15:52:43 +00:00
a3209a2b27
Release buffer pool in Vp8DecoderImpl::Release().
...
Permits reusing an external VP8DecoderImpl instance from another
VideoReceiveStream without a thread-checker DCHECK blowing up. Also
releases buffers that would've been kept in memory even though the
decoder isn't configured.
BUG=
R=magjed@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50449004
Cr-Commit-Position: refs/heads/master@{#8807}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8807 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 13:36:25 +00:00
8904290aca
Make screenshare target bitrate experiment always on
...
BUG=4083
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44699004
Patch from sprang@webrtc.org <sprang@webrtc.org >.
Cr-Commit-Position: refs/heads/master@{#8806}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8806 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 12:50:34 +00:00
9f9ea7e5ab
Clean up webrtc external capture.
...
This cl removes the dependency to the external capture module if external capturing is used in webrtc.
It also removes two external capture methods that is not needed.
Further more it adds I420VideoFrame::Create that takes a pointer to packed memory as input.
R=magjed@webrtc.org , mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43879004
Cr-Commit-Position: refs/heads/master@{#8804}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8804 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 10:55:39 +00:00
443ad403f5
Remove FullStackTest frame pointer handles.
...
Simplifies code, speculative fix for a DCHECK crash in ForemanCifPlr5.
BUG=4451
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45809005
Cr-Commit-Position: refs/heads/master@{#8803}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8803 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 07:34:38 +00:00
6231fb6dac
Prevent crashes when copying a zero-size frame.
...
BUG=4451
R=magjed@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44749004
Cr-Commit-Position: refs/heads/master@{#8802}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8802 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 07:33:11 +00:00
6069032ebb
Refactor audio_coding/isac: removed usage of macro WEBRTC_SPL_LSHIFT_W32
...
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
It is a trivial operation that need no macro. In fact it may be confusing for to the user, since it can be interpreted as having an implicit cast to int32_t.
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44659004
Cr-Commit-Position: refs/heads/master@{#8801}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8801 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 07:03:41 +00:00
4ab23d0e8f
Refactor audio_coding/ilbc: removes usage of macro WEBRTC_SPL_LSHIFT_W32
...
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
It is a trivial operation that need no macro. In fact it may be confusing for to the user, since it can be interpreted as having an implicit cast to int32_t.
Also removes unnecessary casts to int32_t from int16_t.
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48519004
Cr-Commit-Position: refs/heads/master@{#8800}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8800 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 06:01:43 +00:00
bd8c865f43
Remove build-time beamformer flags.
...
RealFourier is now unconditionally enabled since we can fall back to the
Ooura FFT. We no longer need to condition users on rtc_use_openmax_dl.
R=aluebs@webrtc.org , mgraczyk@google.com
Review URL: https://webrtc-codereview.appspot.com/50439004
Cr-Commit-Position: refs/heads/master@{#8799}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8799 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 00:28:42 +00:00
04c50981f8
Add the Ooura FFT to RealFourier.
...
We are using the Ooura FFT in a few places:
- AGC
- Transient suppression
- Noise suppression
The optimized OpenMAX DL FFT is considerably faster, but currently does
not compile everywhere, notably on iOS. This change will allow us to use
Openmax when possible and otherwise fall back to Ooura.
(Unfortunately, noise suppression won't be able to take advantage of it
since it's not C++. Upgrade time?)
R=aluebs@webrtc.org , mgraczyk@chromium.org
Review URL: https://webrtc-codereview.appspot.com/45789004
Cr-Commit-Position: refs/heads/master@{#8798}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8798 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-19 20:07:43 +00:00
80d9aeeda5
Adds full-duplex unit test to AudioDeviceTest on Android
...
BUG=NONE
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42709004
Cr-Commit-Position: refs/heads/master@{#8795}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8795 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-19 15:28:42 +00:00
361981faa8
Use scoped_ptr for ThreadWrapper::CreateThread.
...
BUG=
R=henrika@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45799004
Cr-Commit-Position: refs/heads/master@{#8794}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8794 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-19 14:45:42 +00:00
27c0be9dfe
Remove ThreadObj #define and kThreadMaxNameLength from thread_wrapper.
...
BUG=
R=hbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47679004
Cr-Commit-Position: refs/heads/master@{#8792}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8792 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-19 14:36:43 +00:00
17c64d1c96
Revert "Put ViEFrameProviderBase::DeliverFrame back in the critical section in ViECapturer::DeliverI420Frame"
...
This reverts commit r8724.
Reason for revert: This was not the cause of the tsan issues.
BUG=1128
R=mflodman@webrtc.org , pbos@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50389004
Cr-Commit-Position: refs/heads/master@{#8790}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8790 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-19 10:58:17 +00:00
c7157da599
Use atomic operations for setting/reading the trace filter.
...
The filter is currently being set and read by a number of threads and tripping up tsan.
Original review: https://webrtc-codereview.appspot.com/47609004/
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47659004
Cr-Commit-Position: refs/heads/master@{#8789}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8789 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-19 09:30:45 +00:00
9afaee74ab
Reland 8749: AudioEncoder: return EncodedInfo from Encode() and EncodeInternal()
...
Old review at:
https://webrtc-codereview.appspot.com/43839004/
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45769004
Cr-Commit-Position: refs/heads/master@{#8788}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8788 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-19 08:51:20 +00:00
d21406d333
Remove command-line tool 'video_coding_test'.
...
Removes a lot of code that prevents refactoring VideoCodingModule. Tests
covering the module should be TEST_Fs, and this looks like like fairly
unused code in general.
Adds a 'rtp_player' binary which performs a small subset.
BUG=4391
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44559004
Cr-Commit-Position: refs/heads/master@{#8787}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8787 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-19 08:19:44 +00:00
c4709a2930
Split C++ class from macro overrides to fix Chromium build
...
BUG=chromium:468375
TBR=kjellander@webrtc.org ,ajm@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51409004
Cr-Commit-Position: refs/heads/master@{#8786}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8786 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-19 07:26:21 +00:00
5506a93efd
Expose ViECaptureImpl::DisconnectCaptureDevice() to JNI of WebRTCDemo and call it before releasing camera to deregister the corresponding framecallback. Also stop camera after stop remote rendering as the correct termination order.
...
BUG=4448
TEST=Manual Test
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46649004
Cr-Commit-Position: refs/heads/master@{#8785}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8785 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-19 00:12:40 +00:00
2a8a46dacb
vp8: Add missing call to SetUsageMessage().
...
Without it vp8_coder --help does not work.
BUG=None
TEST=ninja -C out/Debug && out/Debug/vp8_coder --help now shows the
usage message.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44649005
Patch from Thiago Farina <tfarina@chromium.org >.
Cr-Commit-Position: refs/heads/master@{#8783}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8783 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 21:09:16 +00:00
8f76cd25ec
Renaming neteq_opus_fec_quality_test.
...
neteq_opus_fec_quality_test has been modified to test more configurations of Opus than only FEC. It makes sense to rename it to neteq_opus_quality_test. This was planned in
https://webrtc-codereview.appspot.com/45619004/
but was forgotten. This CL handles it, and makes it easy for review.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45709004
Cr-Commit-Position: refs/heads/master@{#8782}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8782 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 20:44:26 +00:00
143451d259
Base start bitrate on last observed bitrate.
...
Instead of setting bitrates based on codec target settings (which may
have previously been capped by a codec max bitrate), fetch the last
bandwidth allocated for this channel. This fixes broken low start bitrates
due to QCIF being set as default codec in WebRtcVideoEngine2 which caps
the max bitrate to 200kbps.
BUG=1788
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43789004
Cr-Commit-Position: refs/heads/master@{#8780}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8780 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 14:40:52 +00:00
5a477a0bc6
DCHECK frame parameters instead of return codes.
...
We should never be creating video frames without width/height. If these
DCHECKs fire we should be fixing the calling code instead.
BUG=4359
R=magjed@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46639004
Cr-Commit-Position: refs/heads/master@{#8779}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8779 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 14:12:38 +00:00
4346d92578
Use SendTimeHistory to keep track of send times in simulations.
...
Use SendTimeHistory to keep track of send times in simulations.
Keep piggybacking send time in PacketInfo for now but use history in
order to be more in line with what we expect to do.
Landing this for sprang@. Original CL: https://review.webrtc.org/43559004/
TBR=sprang@webrtc.org
BUG=4308
Review URL: https://webrtc-codereview.appspot.com/48569004
Cr-Commit-Position: refs/heads/master@{#8778}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8778 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 13:42:48 +00:00
f18993323d
Removing henrik.lundin from OWNERS in video_coding/*
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45699004
Cr-Commit-Position: refs/heads/master@{#8777}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8777 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 09:56:21 +00:00
af612d5e07
Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
...
Original cl description:
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.
With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame
This cl was previously reverted in https://webrtc-codereview.appspot.com/46549004/ .
Patchset 1 contains the original patch after rebase.
Patshet 2 fix webrtc_perf_tests reported in chromium:465306
Note that chromium:465287 is being fixed in https://webrtc-codereview.appspot.com/43829004/
BUG=1128
R=magjed@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47629004
Cr-Commit-Position: refs/heads/master@{#8776}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8776 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 09:51:44 +00:00
6dba1ebd14
Make AudioDecoder stateless
...
The channels_ member varable is removed from the base class, and the
associated accessor function is changed to Channels() which is a pure
virtual function.
R=jmarusic@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43779004
Cr-Commit-Position: refs/heads/master@{#8775}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8775 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 09:48:12 +00:00
fc562e0a56
Delete ACMGenericCodec::Encode and use AudioEncoder::Encode directly
...
Move timestamp conversion out of ACMGenericCodec. Also remove lock
from ACMGenericCodec since the instance is always protected by
acm_crit_sect_ in AudioCodingModuleImpl.
Restructuring the code in AudioCodingModuleImpl::Encode to streamline
the use of locks.
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46479004
Cr-Commit-Position: refs/heads/master@{#8773}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8773 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 07:32:41 +00:00
019955d770
Revert 8749 "We changed Encode() and EncodeInternal() return typ..."
...
The reason is that this cl adds a static initializer so we can't roll webrtc into Chromium.
See audio_encoder.cc and 'sizes' regression here:
http://build.chromium.org/p/chromium/builders/Linux%20x64/builds/186
> We changed Encode() and EncodeInternal() return type from bool to void in this issue:
> https://webrtc-codereview.appspot.com/38279004/
> Now we don't have to pass EncodedInfo as output parameter, but can return it instead. This also adds the benefit of making clear that EncodeInternal() needs to fill in this info.
>
> R=kwiberg@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/43839004
TBR=jmarusic@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49449004
Cr-Commit-Position: refs/heads/master@{#8772}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8772 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 06:38:40 +00:00
edd517bca1
Fix FYI build - add a missing include to event_tracer.h in system_wrappers.
...
TBR=magjed@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/48559004
Cr-Commit-Position: refs/heads/master@{#8768}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8768 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 22:15:28 +00:00
54d072ea20
Add CVO support to video_coding layer.
...
CVO is, instead of rotating frame on the capture side, to have renderer rotate the frame based on a new rtp header extension.
The change includes
1. encoder side needs to pass this from raw frame to the encoded frame.
2. decoder needs to copy it from rtp packet (only the last packet of a frame has this info) to decoded frame.
R=mflodman@webrtc.org
TBR=stefan@webrtc.org
BUG=4145
Review URL: https://webrtc-codereview.appspot.com/46429006
Cr-Commit-Position: refs/heads/master@{#8767}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8767 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 21:55:37 +00:00
63a10978e1
Remove troublesome Windows line ending.
...
R=decurtis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48549004
Cr-Commit-Position: refs/heads/master@{#8766}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8766 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 21:50:29 +00:00
462dbcfc2a
Fix bug in Transport where channel_.clear() was being called without a lock.
...
Looks like this snuck in between misaligned braces.
Also switching to C++11 for loops, reducing lock scopes in a few places and removing locks in others.
BUG=4444
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43769004
Cr-Commit-Position: refs/heads/master@{#8765}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8765 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 21:40:26 +00:00
b493cb4497
Add storage alignment fix for opengles2.0 for iOS
...
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40179004
Patch from Iurii Shevchuk <youwrk@gmail.com >.
Cr-Commit-Position: refs/heads/master@{#8764}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8764 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 20:18:42 +00:00
da4fcc494c
Add minor fixes to video_capture_ios.mm in order to make it more robust.
...
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46429005
Patch from Iurii Shevchuk <youwrk@gmail.com >.
Cr-Commit-Position: refs/heads/master@{#8763}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8763 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 20:13:49 +00:00
779c3d16b9
Use ByteReader/ByteWriter instead of rtputility and manual shift/add.
...
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41289004
Cr-Commit-Position: refs/heads/master@{#8761}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8761 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 16:44:54 +00:00
09098dabd3
Fix screenshare loopback target bitrate which isn't correctly configured
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BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48539004
Cr-Commit-Position: refs/heads/master@{#8760}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8760 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 16:28:11 +00:00
25819b8294
Revert 8753 "Use atomic operations for setting/reading the trace..."
...
Caused VP9 test to fail on TSAN and doesn't build in some configuration due to
"../webrtc/base/criticalsection.h:181:12: error: cannot compile this atomic library call yet"
:-(
> Use atomic operations for setting/reading the trace filter.
> The filter is currently being set and read by a number of threads and tripping up tsan.
>
> R=mflodman@webrtc.org
> BUG=
>
> Review URL: https://webrtc-codereview.appspot.com/47609004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51369004
Cr-Commit-Position: refs/heads/master@{#8759}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8759 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 15:35:41 +00:00
b91d0f5130
1. Have IPIsPrivate calling IPIsLinkLocal
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2. Also check the Mac based IPv6
3. move the ip filtering into createnetwork. It shouldn't be done in IsIgnoredNetwork as the IP inside that could change later.
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48509004
Cr-Commit-Position: refs/heads/master@{#8758}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8758 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 14:43:42 +00:00
3093390479
Parsing of transport wide sequence number rtp extension header.
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Plus some refactoring to correctly handle padding.
BUG=4311
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45429004
Cr-Commit-Position: refs/heads/master@{#8757}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8757 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 14:33:46 +00:00
7c64ed2e0c
Move trace_event and associated files to webrtc/base.
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Also starting to use TRACE_EVENT from thread.cc in webrtc/base, to track Invoke() calls.
BUG=
R=magjed@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42769004
Cr-Commit-Position: refs/heads/master@{#8755}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8755 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 14:26:15 +00:00
7c112f3e5a
Adding build_opus as a switch in GYP.
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This is to allow not building Opus. On non-chromium non-gyp chases, one can let WebRTC depend on other Opus builds.
BUG=
R=kjellander@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43739004
Cr-Commit-Position: refs/heads/master@{#8754}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8754 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 14:05:18 +00:00
c383c24c2b
Use atomic operations for setting/reading the trace filter.
...
The filter is currently being set and read by a number of threads and tripping up tsan.
R=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/47609004
Cr-Commit-Position: refs/heads/master@{#8753}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8753 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 13:47:16 +00:00
a846371ace
Modify EventPosix to prevent spurious wakeups.
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pthread_cond_{timedwait,wait} are allowed to spuriously wake up as if
they were signaled. To prevent this being interpreted as a "real"
signaling of the event (ThreadWrapper for instance depends on it being
an actual signal) we need to check whether the event was actually
signalled or not.
BUG=4413
R=andresp@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49369004
Cr-Commit-Position: refs/heads/master@{#8752}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8752 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 13:14:46 +00:00
a78a94e838
Fix RateTracker to set an initial reference time when first updated.
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BUG=4442
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43829004
Cr-Commit-Position: refs/heads/master@{#8751}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8751 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 12:45:41 +00:00
