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Author SHA1 Message Date
tommi@webrtc.org
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
braveyao@webrtc.org
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
tkchin@webrtc.org
8cc47e926c Objective-C readability review. 
BUG=
R=rsesek@chromium.org

Review URL: https://webrtc-codereview.appspot.com/34679004

Cr-Commit-Position: refs/heads/master@{#8784}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8784 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 23:38:45 +00:00
kjellander@webrtc.org
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
minyue@webrtc.org
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
guoweis@webrtc.org
840da7b755 Implement Rotation in Android Renderer. 
Make use of rotation information from the frame and rotate it accordingly when we render the frame.

BUG=4145
R=glaznev@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8770

Review URL: https://webrtc-codereview.appspot.com/50369004

Cr-Commit-Position: refs/heads/master@{#8781}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8781 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 16:58:49 +00:00
pbos@webrtc.org
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
pbos@webrtc.org
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
stefan@webrtc.org
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
henrik.lundin@webrtc.org
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
perkj@webrtc.org
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
henrik.lundin@webrtc.org
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
magjed@webrtc.org
14ee8cc9c7 WebRtcVideoFrame: Support odd resolutions 
We currently truncate the resolution of frames to a multiple of 4. This is unnecessary as everything supports odd resolutions now.

R=fbarchard@google.com, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43819004

Cr-Commit-Position: refs/heads/master@{#8774}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8774 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 09:22:19 +00:00
henrik.lundin@webrtc.org
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
tommi@webrtc.org
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
guoweis@webrtc.org
3fffd66dfa Revert "Implement Rotation in Android Renderer." 
This reverts commit 835ec63d8a64bbc8a573a5e0b7a09659188122d2.

TBR=guoweis@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/51399004

Cr-Commit-Position: refs/heads/master@{#8771}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8771 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 04:20:47 +00:00
guoweis@webrtc.org
835ec63d8a Implement Rotation in Android Renderer. 
Make use of rotation information from the frame and rotate it accordingly when we render the frame.

BUG=4145
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50369004

Cr-Commit-Position: refs/heads/master@{#8770}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8770 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 02:44:39 +00:00
pthatcher@webrtc.org
52cd828e17 Allow webrtc external encoder factories to declare encoders have internal camera sources. 
This flag is passed to existing VieExternalCodec API (and others) to denote encoders that don't require/expect frames from the normal capture pipeline. This is the simplest way to allow camera->encoder texture support, until textures are supported through the normal camera pipeline and the lifetime issues are all figured out (I hear this is on the backlog, but not there yet).

Ideally, the flag would be on the encoder, but that doesn't work with SimulcastEncoderAdapter, since it doesn't create an encoder right away.

Note that this change only affects WebRtcVideoEngine (not WRVE2), since WRVE2 uses video_send_stream, and my hope is that by the time things have switched to WRVE2, textures will be supported with the normal camera pipeline and the dependency on internal sources can be thrown away.

BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42349004

Cr-Commit-Position: refs/heads/master@{#8769}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8769 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 02:25:18 +00:00
tommi@webrtc.org
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
guoweis@webrtc.org
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
pthatcher@webrtc.org
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
tommi@webrtc.org
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
tkchin@webrtc.org
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
tkchin@webrtc.org
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
glaznev@webrtc.org
2161234cf6 Add new features to AppRTCDemo from private repo. 
- Add HUD fragment with HUD related controls and more
HUD statistics.
- Create and set all peer connection constraints in
PeerConnectionClient class.
- Handle registration request in web socket class internally
once web socket connection is opened.

R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44669004

Cr-Commit-Position: refs/heads/master@{#8762}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8762 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 18:24:19 +00:00
sprang@webrtc.org
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
sprang@webrtc.org
09098dabd3 Fix screenshare loopback target bitrate which isn't correctly configured 
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
tommi@webrtc.org
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
guoweis@webrtc.org
b91d0f5130 1. Have IPIsPrivate calling IPIsLinkLocal 
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
sprang@webrtc.org
3093390479 Parsing of transport wide sequence number rtp extension header. 
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
kjellander@webrtc.org
1e6925274a Write commit position as a comment in Chromium DEPS. 
This will make it easier to track which revision is
in a certain Chrome release, since you don't have to
look up the Git hash every time.

Also rename svn_revision to commit_position to make
it more clear what the number is in the post-SVN era.

TESTED=tools/autoroller/roll_webrtc_in_chromium.py --chromium-checkout /ssd/chrome/src --verbose --ignore-checks --dry-run --close-previous-roll
and verified in the modified DEPS file that the comment was set.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49439004

Cr-Commit-Position: refs/heads/master@{#8756}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8756 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 14:30:22 +00:00
tommi@webrtc.org
7c64ed2e0c Move trace_event and associated files to webrtc/base. 
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
minyue@webrtc.org
7c112f3e5a Adding build_opus as a switch in GYP. 
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
tommi@webrtc.org
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
pbos@webrtc.org
a846371ace Modify EventPosix to prevent spurious wakeups. 
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
perkj@webrtc.org
a78a94e838 Fix RateTracker to set an initial reference time when first updated. 
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
magjed@webrtc.org
e155dbeae9 VP8/9EncoderImpl::Encode: Check resolution of input I420VideoFrame 
This CL adds checks in Encode to guard against memory reads out of bounds.

R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46429008

Cr-Commit-Position: refs/heads/master@{#8750}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8750 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 12:27:40 +00:00
jmarusic@webrtc.org
0cb612b43b 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

Cr-Commit-Position: refs/heads/master@{#8749}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8749 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 12:13:13 +00:00
magjed@webrtc.org
73d763e71f Add I420 buffer pool to avoid unnecessary allocations 
Now when we don't use SwapFrame consistently anymore, we need to recycle allocations with a buffer pool instead. This CL adds a buffer pool class, and updates the vp8 decoder to use it. If this CL lands successfully I will update the other video producers as well.

BUG=1128
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41189004

Cr-Commit-Position: refs/heads/master@{#8748}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8748 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 11:41:15 +00:00
pbos@webrtc.org
ae222b5be6 Remove dead code in WebRtcVideoEngine2 unittests. 
BUG=1788
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43609004

Cr-Commit-Position: refs/heads/master@{#8747}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8747 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 10:48:28 +00:00
magjed@webrtc.org
858024f1d9 WebRtcVideoFrame: Initialize members in empty constructor 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41319004

Cr-Commit-Position: refs/heads/master@{#8746}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8746 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 08:47:17 +00:00
kjellander@webrtc.org
646eeacf8c Roll chromium_revision 8d51d96..bd49b12 (320682:320783) 
Pulls in new libvpx version that allows us to re-enable the
VideoProcessorIntegrationTest.ProcessNoLossDenoiserOnVP9
test in webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc

Relevant changes:
* src/third_party/libvpx: 763fe7a..f80cf58
* src/tools/gyp: 4a9b712..d174d75
Details: 8d51d96..bd49b12/DEPS

Clang version was not updated in this roll.

BUG=4418
TBR=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41339004

Cr-Commit-Position: refs/heads/master@{#8745}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8745 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 08:26:17 +00:00
marpan@webrtc.org
06d93909cd Adjust a threshold in VP9 test. 
For upcoming libvpx roll.

TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/43799004

Cr-Commit-Position: refs/heads/master@{#8744}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8744 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 22:13:16 +00:00
pthatcher@webrtc.org
592470b4ff Remove a dependency of BaseChannel on WebRtcSession by having WebRtcSession push down new media descriptions to BaseChannel rather than having BaseChannel listen to the description changes from WebRtcSession. 
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47599004

Cr-Commit-Position: refs/heads/master@{#8743}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8743 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 21:16:23 +00:00
kjellander@webrtc.org
12e7951bf2 Remove libvpx suppression due to fixed bug. 
BUG=webm:962
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45719004

Cr-Commit-Position: refs/heads/master@{#8742}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8742 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 20:43:47 +00:00
pthatcher@webrtc.org
6ad507ac35 Refactor how the TransportChannels are set in the BaseChannel to rely lesson Session, so that in the future we can rely on Transport instead, and also be able to change Transports on the fly for BUNDLE. 
Also, remove channel_name.  It's no longer needed.

This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=decurtis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43719004

Cr-Commit-Position: refs/heads/master@{#8741}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8741 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 20:19:42 +00:00
pthatcher@webrtc.org
4eeef584a7 Remove a hacky dependency of BaseChannel on BaseSession by moving the handling of DTLS setup failure into a signal on BaseChannel rather than a method call on BaseSession. 
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=decurtis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47589004

Cr-Commit-Position: refs/heads/master@{#8740}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8740 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 19:34:40 +00:00
pthatcher@webrtc.org
c04a97f054 Move from BaseSession::GetStats to WebRtcSession::GetTransportStats 
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

Review URL: https://webrtc-codereview.appspot.com/45639004

Cr-Commit-Position: refs/heads/master@{#8739}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 19:32:23 +00:00
tommi@webrtc.org
aba9219e5c Change ThreadPosix to use an auto-reset event instead of manual reset now that we know the problem we had with EventWrapper::Wait was simply a bug in the EventWrapper. Also removing |started_| since we can just check the thread_ instead. 
R=pbos@webrtc.org
BUG=4413

Review URL: https://webrtc-codereview.appspot.com/47539004

Cr-Commit-Position: refs/heads/master@{#8738}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 16:06:16 +00:00
henrik.lundin@webrtc.org
02d166b735 Fixing a race condition in ACMGenericCodec 
The old object was deleted before the pointer to it was removed from
the decoder proxy.

BUG=chromium:467209
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49429004

Cr-Commit-Position: refs/heads/master@{#8736}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 14:33:43 +00:00