Commit Graph

3687 Commits

Author SHA1 Message Date
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
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
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.
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.
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.
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.
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.
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
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
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
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
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
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
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
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
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
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
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
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
8bd2f40a8c Remove code related to REMB suppressor experiment.
Stats indicate this isn't helping. Ditching the whole thing.

BUG=4082
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47569004

Cr-Commit-Position: refs/heads/master@{#8734}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8734 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 14:11:42 +00:00
2056ee3e3c Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."
This reverts commit r8731.

Reason for revert: Breakes Chromium FYI bots.

TBR=hbos, tommi

Review URL: https://webrtc-codereview.appspot.com/40359004

Cr-Commit-Position: refs/heads/master@{#8733}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8733 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 13:48:18 +00:00
93d9d6503e I420VideoFrame.CreateFrame: Removed unnecessary buffer size arguments.
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45629004

Cr-Commit-Position: refs/heads/master@{#8732}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8732 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 13:26:41 +00:00
2dc5fa69b2 Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*.
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40299004

Cr-Commit-Position: refs/heads/master@{#8731}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8731 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 13:02:19 +00:00
7f7d7e3427 Prevent crash in NetEQ when decoder overflow.
NetEQ can crash when decoder gives too many output samples than it can handle. A practical case this happens is when multiple opus packets are combined.

The best solution is to pass the max size to the ACM decode function and let it return a failure if the max size if too small.

BUG=4361
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45619004

Cr-Commit-Position: refs/heads/master@{#8730}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8730 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 12:31:19 +00:00
eed2fcaa76 Roll chromium_revision 00e438c..8d51d96 (320241:320682)
Relevant changes:
* src/third_party/android_tools: fd5a8ec..98a4345
Details: 00e438c..8d51d96/DEPS

This required updating our Android projects to API level 22,
as third_party/android_tools dropped support for API level 21.

Command used:
perl -pi -e "s/android-21/android-22/g" `find . -name project.properties`
Using 'android update project' would also work but that changes the
ANDROID_SDK_ROOT -> ANDROID_HOME, which the Chromium build toolchain
doesn't set properly when build/android/envsetup.sh is sourced.

Clang version was not updated in this roll.

R=henrika@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42779004

Cr-Commit-Position: refs/heads/master@{#8728}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8728 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 09:00:41 +00:00
6107ba12f9 Put ViEFrameProviderBase::DeliverFrame back in the critical section in ViECapturer::DeliverI420Frame
The CL that moved it out of the critical section is here: https://webrtc-codereview.appspot.com/43669004/

BUG=1128
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45679005

Cr-Commit-Position: refs/heads/master@{#8724}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8724 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-14 11:50:14 +00:00