2977 Commits

Author SHA1 Message Date
asapersson@webrtc.org
83b5200f95 Add framerate for complete received frames to histogram stats:
"WebRTC.Video.CompleteFramesReceivedPerSecond".

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7762 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-28 10:17:13 +00:00
aluebs@webrtc.org
cc144deaab Make bands vector in SplittingFilter Analysis const
BUG=webrtc:3146
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7761 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-28 00:26:27 +00:00
aluebs@webrtc.org
8789376cd3 Move ChannelBuffer class to channel_buffer file
No change in functionallity.

BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7760 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-27 23:40:25 +00:00
pbos@webrtc.org
d87213af49 Remove unused RtpStatistics struct.
This unused struct is basically a copy of RtcpStatistics in
webrtc/common_types.h. I expect this wasn't properly removed when that
one was added.

R=tommi@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/25239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7758 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-27 13:48:35 +00:00
asapersson@webrtc.org
d952c40c7e Add receive bitrates to histogram stats:
- total bitrate ("WebRTC.Video.BitrateReceivedInKbps")
- media bitrate ("WebRTC.Video.MediaBitrateReceivedInKbps")
- rtx bitrate ("WebRTC.Video.RtxBitrateReceivedInKbps")
- padding bitrate ("WebRTC.Video.PaddingBitrateReceivedInKbps")

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27189005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7756 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-27 07:38:56 +00:00
aluebs@webrtc.org
79b9eba3ab Implement 3 band splitting filter bank by upsampling and splitting twice into 2 bands
Implemented the 3 bands splitting filter bank by:
1. Upsample by 4/3.
2. Split twice into 2 bands.
3. Discard upper most band, because it is empty anyway.

A unittest was also implemented:
1. Generate a signal from presence or absence of sine waves of different frequencies.
2. Split into 3 bands and check their presence or absence.
3. Recombine the bands.
4. Calculate delay (as it is an IIR it depends on frequency).
5. Check that the cross correlation of input and output is high enough at that delay.

BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7754 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-26 20:21:38 +00:00
jiayl@webrtc.org
7806d8fe40 Fix an ASSERT that fires in a browser test for renegotiation.
See https://code.google.com/p/chromium/issues/detail?id=293125#c33

BUG=crbug/293125
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7753 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-26 19:58:50 +00:00
andrew@webrtc.org
a56a2c57cf Enabling building with NEON on ARM64
This patch enables NEON on ARM64 platform. Passed building both on
Android ARMv7 and Android ARM64.

BUG=3580
R=andrew@webrtc.org, jridges@masque.com

Review URL: https://webrtc-codereview.appspot.com/25069004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7751 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-26 17:01:40 +00:00
henrik.lundin@webrtc.org
91d928e737 Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader
This is in preparation for creating a new class RtpFileWriter which
will use the same RtpPacket struct.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7749 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-26 15:50:30 +00:00
pbos@webrtc.org
008731868a Implement settable min/start/max bitrates in Call.
These parameters are set by the x-google-*-bitrate SDP parameters. This
is implemented on a Call level instead of per-stream like the currently
underlying VideoEngine implementation to allow this refactoring to not
reconfigure the VideoCodec at all but rather adjust bandwidth-estimator
parameters.
Also implements SetMaxSendBandwidth in WebRtcVideoEngine2 as it's a SDP
parameter and allowing it to be dynamically readjusted in Call.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/26199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7746 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-25 14:03:34 +00:00
pbos@webrtc.org
b951eb12c9 Add back EXPECT_TRUEs.
These shouldn't fail, but EXPECT_TRUE gives nicer error messages that
work in Release. These changes got through unreviewed in r7726.

R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/26249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7745 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-25 11:13:28 +00:00
pbos@webrtc.org
ba253473da Reenable GetStats test.
Also increasing start bitrate to have the test go significantly faster
on average. Hopefully an assert hit in the jitter buffer while running
this test was fixed in r7735.

R=stefan@webrtc.org
BUG=4014

Review URL: https://webrtc-codereview.appspot.com/26239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7744 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-25 09:39:04 +00:00
henrik.lundin@webrtc.org
03499a0e95 Add wav output capability to neteq_rtpplay
This CL makes neteq_rtpplay capable of writing to wav files as well as
pcm files. This is done through the new class OutputWavFile, which
wraps a WavWriter object in an AudioSink interface.

BUG=2692
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7740 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-24 14:50:53 +00:00
henrik.lundin@webrtc.org
aff1751c96 Add new test for VP8 packetizer to test tight partitions
It was discovered that if remaining_bytes is an exact multiple of
max_payload_len in RtpPacketizerVp8::CalcNextSize, then the packetizer
will produce too many packets (i.e., split the payload into more
packets than needed).

This CL adds a test to trigger the problem.

BUG=4019
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7739 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-24 12:36:58 +00:00
kjellander@webrtc.org
8562f23acb OWNERS: Remove tomasl@ and mallinath@
mallinath@ has left the team and tomasl@ says he doesn't
know why he's owner in webrtc/test/channel_transport

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

Review URL: https://webrtc-codereview.appspot.com/30119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7736 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-24 10:05:05 +00:00
pbos@webrtc.org
4f16c874c6 Simplifying VideoReceiver and JitterBuffer.
Removing frame_buffers_ array and dual-receiver mechanism. Also adding
some thread annotations to VCMJitterBuffer.

R=stefan@webrtc.org
BUG=4014

Review URL: https://webrtc-codereview.appspot.com/27239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7735 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-24 09:06:48 +00:00
pbos@webrtc.org
9334ac2d78 Use vector of CSRCs for DeliverFrame & SetCSRCs.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28029004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7734 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-24 08:25:50 +00:00
andrew@webrtc.org
1153322cf8 Build fix for MIPS Android Webview build.
Excluding optimizations to support MIPS32R6 platform for Android Webview build (see also https://code.google.com/p/webrtc/source/detail?r=7580).

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7729 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-21 16:28:32 +00:00
kjellander@webrtc.org
ad0e71c9a3 Update mock_frame_dropper.h to use size_t
This mock was missed in the work of
https://webrtc-codereview.appspot.com/23129004 since the file
is not currently used by any test in this repo.

BUG=chromium:81439
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7727 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-21 09:40:57 +00:00
pkasting@chromium.org
4591fbd09f Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
henrik.lundin@webrtc.org
6ff3ac1db8 Fix problems if first packet into NetEq is rejected
This CL fixes the problem described in issue 4021. In summary, of the
very first packet coming in to NetEq gets rejected because the RTP
payload type is unknown, subsequent GetAudio calls would trigger asserts
(in debug builds). The problem was that the first_packet_ variable was
reset and new_codec_ was set, even though the packet was discarded
further down the line. Now, these variables are modified after the
packet has been verified.

BUG=4021
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7724 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 14:14:49 +00:00
henrik.lundin@webrtc.org
ed91068bf1 Create a NetEq test for when the first incoming payload type is unknown
This test is currently disabled. It triggers an issue where the NetEq
will trigger asserts on subsequent GetAudio calls if the first inserted
packet is rejected, for instance since the payload type is unknown to
NetEq.

BUG=4021
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7723 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 11:01:02 +00:00
asapersson@webrtc.org
049e4ece30 Change default values for CpuOveruseOptions.
Enabled method based on encode time and modified values for the low (60->55) and high threshold (90->85).

Moved DelayedEncoder to fake_encoder.h and added configuration for the delay.

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

Review URL: https://webrtc-codereview.appspot.com/30969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7722 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 10:19:46 +00:00
henrik.lundin@webrtc.org
40af3a56e4 Revert "Add DCHECK to ensure that NetEq's packet buffer is not empty"
This reverts r7719. It failed to compile in Chromium.

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7720 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-19 13:46:52 +00:00
henrik.lundin@webrtc.org
6f6ef72950 Add DCHECK to ensure that NetEq's packet buffer is not empty
This DCHECK ensures that one packet was inserted after the buffer was
flushed.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7719 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-19 13:02:24 +00:00
aluebs@webrtc.org
087da13fe8 Add empty 3 band splitting filter API
This is only an empty API that will never be used. For now is 48kHz not supported in AudioProcessing. For that it needs to be added in InitializeLocked. But before the 3 band filter bank needs to be populated.

BUG=webrtc:3146
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7715 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-17 23:01:23 +00:00
pkasting@chromium.org
2656bf813f Fix ExpectedQueueTimeMs() to avoid truncation or overflow.
BUG=none
TEST=none
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7714 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-17 22:21:14 +00:00
guoweis@webrtc.org
930e004a81 Add jmi field for packets discarded due to network error
Also included the total packets attempted to send.

BUG=427555

Copied from https://webrtc-codereview.appspot.com/25959004/

R=harryjin@google.com, juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7693

Review URL: https://webrtc-codereview.appspot.com/32039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7713 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-17 19:42:14 +00:00
pbos@webrtc.org
f5b56fbc41 Annotate COMPILE_ASSERT with __attribute__((unused)).
Also renames UNUSED -> ATTRIBUTE_UNUSED to be able to use this when
building peerconnection_jni.cc which apparently has this defined to
something else.

R=kjellander@webrtc.org
TBR=mflodman@webrtc.org
BUG=4018

Review URL: https://webrtc-codereview.appspot.com/28039005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7711 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-17 13:47:38 +00:00
henrik.lundin@webrtc.org
966a708b93 Use RtpFileSource in NetEqDecodingTest
This CL removes the dependency on the old NETEQTEST_RTPpacket class
from the NetEqDecodingTest code, and also removes the dependency
from the modules_unittests target to neteq_test_tools.

BUG=2692
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7709 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-17 09:08:38 +00:00
henrike@webrtc.org
6a782c2a46 Revert 7693 "Add jmi field for packets discarded due to network error" breaks chromium's webrtc_cases.
TBR=guoweis@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/25179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7706 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-14 22:33:13 +00:00
aluebs@webrtc.org
be05c74ec8 Wrap the splitting filter in its own class
This doesn't change the behavior at all.
The logic behind this is having one class which manages all the splitting filters, because in the future we plan to add a 3 band one for 48kHz support.
It also breaks the dependency of the AudioBuffer with the filter states of these filters (which are going to be different for the 3 band one). The AudioBuffer is complicated enough and is going to need changes to support 3 bands in the future, so any simplification is a good idea.
On top of that it eliminates repeated code in the APM (now only iterating over channels, but then also deciding in how many bands to split). This should be managed by the AudioBuffer directly.

BUG=webrtc:3146
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7705 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-14 22:18:10 +00:00
pbos@webrtc.org
67c22478a4 Disable EndToEnd.GetStats test.
Looks like this test exposes a bug in jitter buffer after enabling
multiple streams. Will disable to be able to debug it in peace and not
have to revert.

TBR=stefan@webrtc.org
BUG=4014

Review URL: https://webrtc-codereview.appspot.com/31009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7704 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-14 17:42:51 +00:00
pbos@webrtc.org
ece3890d3a Report total bitrate for all streams in GetStats.
This regression wasn't caught because I accidentally disabled multiple
streams for EndToEndTest.GetStats in a refactoring.

R=stefan@webrtc.org, xians@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/27179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7701 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-14 11:52:04 +00:00
pbos@webrtc.org
49ff40e32e Make SetREMBData accept vector of SSRCs.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7697 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 14:42:37 +00:00
pbos@webrtc.org
a9c2d454bd Fix and enable CanReceiveFec test.
Test relied on the first protected media packet that was dropped to
actually be rendered, while rendering it could have been skipped on slow
systems due to newer frames being decoded before rendering happens.

R=stefan@webrtc.org
BUG=3269

Review URL: https://webrtc-codereview.appspot.com/25159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7696 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 14:40:15 +00:00
bjornv@webrtc.org
ee30082af8 Set correct sample rate in far_frame in audioproc tool.
One debug recording with non matching sample rates between render and capture revealed a bug in modules/audio_processing/test/process_test.cc
The far_frame (render audio frame) used was loaded with the capture rate instead of the render rate with a data length mismatch error as result.

BUG=N/A
TESTED=manually on linux
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7695 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 11:00:10 +00:00
kjellander@webrtc.org
52bb521b47 Update isolate files for Android APK tests.
This should speed up test execution on Android since only
the files needed by the test will be processed (instead
of the whole data + resources directories).

A few files for modules_unittests had to be explicitly added
for Android, since they were previously a part of the
add-whole-directories entries for the resources and data
directories.

BUG=webrtc:3741
TEST=Passing android+android_rel trybots.
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7694 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 08:35:05 +00:00
guoweis@webrtc.org
312614a438 Add jmi field for packets discarded due to network error
Also included the total packets attempted to send.

BUG=427555

Copied from https://webrtc-codereview.appspot.com/25959004/

R=harryjin@google.com, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7693 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 03:38:05 +00:00
jiayl@webrtc.org
90b9b08332 Fix a platform check to use WEBRTC_WIN instead of OS_WIN.
BUG=4006
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/25169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7691 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-12 20:53:00 +00:00
magjed@webrtc.org
ea73ff7267 webrtc::Scaler: Preserve aspect ratio
BUG=3936
R=glaznev@webrtc.org, stefan@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7679

Review URL: https://webrtc-codereview.appspot.com/28969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7689 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-12 09:52:03 +00:00
magjed@webrtc.org
0b3d89b500 VideoSendStreamTest.SwapsI420VideoFrames: Initialize frame memory to avoid drmemory errors
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7688 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-12 08:58:49 +00:00
kjellander@webrtc.org
14ea50a8e3 Change the static_library("webrtc") to a source set in the GN build.
Static libraries cannot have only headers as sources (libtool complains
that there's nothing to actually link).

TBR=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/27159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7687 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-12 07:56:21 +00:00
andrew@webrtc.org
0e37b898f0 replace inline assembly WebRtcAecm_CalcLinearEnergiesNeon by intrinsics.
The modification only uses the unique part of the CalcLinearEnergies
 function. Pass byte to byte conformance test both on ARMv7 and ARM64,
 and the single function performance is similar with original assembly
 version on different platforms. If not specified, the code is compiled
 by GCC 4.6. The result is the "X version / C version" ratio, and the
 less is better.

| run 100k times             | cortex-a7 | cortex-a9 | cortex-a15 |
| use C as the base on each  |  (1.2Ghz) |  (1.0Ghz) |   (1.7Ghz) |
| CPU target                 |           |           |            |
|----------------------------+-----------+-----------+------------|
| Neon asm                   |    19.48% |    19.26% |     13.68% |
| Neon inline                |    27.90% |    28.87% |     17.79% |
| Neon intrinsics (GCC 4.8)  |    18.69% |    20.18% |     14.69% |
| Neon intrinsics (LLVM 3.4) |    18.52% |    21.15% |     13.56% |

BUG=3580
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23349004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7686 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 19:34:14 +00:00
andrew@webrtc.org
e497be3de1 replace inline assembly WebRtcAecm_StoreAdaptiveChannelNeon by intrinsics.
The modification only uses the unique part of the StoreAdaptiveChannel
 function. Pass byte to byte conformance test both on ARM32 and ARM64,
 and the single function performance is similar with original assembly
 version on different platforms. If not specified, the code is compiled
 by GCC 4.6.  The result is the "X version / C version" ratio, and the
 less is better.

| run 100k times             | cortex-a7 | cortex-a9 | cortex-a15 |
| use C as the base on each  |  (1.2Ghz) |  (1.0Ghz) |   (1.7Ghz) |
| CPU target                 |           |           |            |
|----------------------------+-----------+-----------+------------|
| Neon asm                   |    20.97% |    37.70% |     25.41% |
| Neon inline                |    36.93% |    51.80% |     38.14% |
| Neon intrinsics (GCC 4.6)  |    27.78% |    43.71% |     26.50% |
| Neon intrinsics (GCC 4.8)  |    27.16% |    38.22% |     26.87% |
| Neon intrinsics (LLVM 3.4) |    27.82% |    39.90% |     26.69% |

Change-Id: Ia55d8a268a70164b50676c604ae40b68fc183106

BUG=3580
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30029004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7685 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 19:32:33 +00:00
jiayl@webrtc.org
0e71070207 Use ScreenCapturer to capture the whole and clip to the window rect when the shared window is on the top.
BUG=crbug/403703, crbug/316603
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/22759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7684 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 18:15:55 +00:00
tnakamura@webrtc.org
a367aeab82 Bump to version 40
TBR=niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/26109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7683 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 16:23:15 +00:00
magjed@webrtc.org
f7c5d4fac7 Revert 7679 "webrtc::Scaler: Preserve aspect ratio"
> webrtc::Scaler: Preserve aspect ratio
> 
> BUG=3936
> R=glaznev@webrtc.org, stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/28969004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7682 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 13:12:09 +00:00
magjed@webrtc.org
809986b95f webrtc::Scaler: Preserve aspect ratio
BUG=3936
R=glaznev@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7679 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 09:51:30 +00:00
asapersson@webrtc.org
cd621a8657 Add thread annotations to overuse_frame_detector class.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7678 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 09:40:19 +00:00