Commit Graph

6910 Commits

Author SHA1 Message Date
marpan@webrtc.org
f6b7c7e6a6 Exclude SendsAndReceivesVP9 for WinDrMemory.
https://code.google.com/p/webm/issues/detail?id=872

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7639 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 05:09:26 +00:00
marpan@webrtc.org
e1745cbb7c Adjust parameter in vp9 rate control test.
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7638 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 02:55:53 +00:00
marpan@webrtc.org
5f1e2e42a8 Increase speed setting for VP9 (from 5 to 6) and re-enable end_to_end test.
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7637 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 02:02:28 +00:00
tkchin@webrtc.org
ee9d61ce45 This fixes a small memory leak (found using Xcode/Instruments on iOS) in
the ObjC bindings of PeerConnection. The generated session description has
to be released by the recipient

BUG=3985
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28959004

Patch from Matthias Liebig <matthias.gcode@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7636 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 22:01:53 +00:00
pbos@webrtc.org
6a364fe11b Remove uses of build date/time.
Uses of __DATE__ and __TIME__ are blocking deterministic Chromium
builds. We're not really making use of these, and if anything they're
likely to be misleading as it's impossible to distinguish between a new
revision and a freshly-built old branch.

R=mflodman@webrtc.org, tnakamura@webrtc.org
BUG=3983

Review URL: https://webrtc-codereview.appspot.com/27039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7635 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 17:40:28 +00:00
stefan@webrtc.org
0bae1fab4a Wire up bandwidth stats to the new API and webrtcvideoengine2.
Adds stats to verify bandwidth and pacer stats.

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

Review URL: https://webrtc-codereview.appspot.com/24969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 14:05:29 +00:00
buildbot@webrtc.org
a22a628356 (Auto)update libjingle 79205306-> 79244016
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7633 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 13:25:48 +00:00
kjellander@webrtc.org
72fd339352 Restore old behavior for Android in fileutils.cc
From r7014 the Android APK tests are designed to be
build from a standalone WebRTC checkout instead of a
Chromium checkout. Because of that, the special handling
for both cases can be removed.

I also don't think we need to use the
base::android::GetExternalStorageDirectory() method since
all devices has a symlink at /sdcard that points
to /storage/emulated/legacy on the Android device.

This essentially reverts the changes in
https://webrtc-codereview.appspot.com/1754005/
plus some minor changes.

BUG=webrtc:3741
TEST=Locally running test_support_unittests APK test on an
Android device using:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py gtest -s test_support_unittests --verbose --isolate-file-path=webrtc/test/test_support_unittests.isolate
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7632 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 06:28:50 +00:00
kjellander@webrtc.org
f6e1600a7d Roll chromium_revision d3db2ff..375f736
Mainly to bring in https://codereview.chromium.org/636083002
updates to opus.

Relevant changes (d3db2ff..375f736/DEPS)
* buildtools: 6bc51af..1ca1f25
* third_party/libsrtp: 98284c8..6446144
* tools/gyp: a61e860..b13d8f2
* tools/swarming_client: bcb3bc3..41036ec

Clang version was not updated in this roll.

BUG=
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/24179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7631 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 02:09:20 +00:00
glaznev@webrtc.org
dc8662435b Fix android_clang build.
BUG=
R=kjellander@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7630 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 01:15:10 +00:00
niklas.enbom@webrtc.org
368215dacb Revert 7623 "Remove the state_ member from AudioDecoder"
Breaks Chrome compile:
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(131) : error C3867: 'webrtc::NetEqImpl::InsertPacketInternal': function call missing argument list; use '&webrtc::NetEqImpl::InsertPacketInternal' to create a pointer to member
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(131) : error C3861: 'LOG_FERR1': identifier not found
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(152) : error C3867: 'webrtc::NetEqImpl::InsertPacketInternal': function call missing argument list; use '&webrtc::NetEqImpl::InsertPacketInternal' to create a pointer to member
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(152) : error C3861: 'LOG_FERR1': identifier not found
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(169) : error C3867: 'webrtc::NetEqImpl::GetAudioInternal': function call missing argument list; use '&webrtc::NetEqImpl::GetAudioInternal' to create a pointer to member
...

> Remove the state_ member from AudioDecoder
> 
> The subclasses that need a state pointer should declare them---with
> the right type, not void*, to get rid of all those casts.
> 
> Two small but not quite trivial cleanups are included because they
> blocked the state_ removal:
> 
>   - AudioDecoderG722Stereo now inherits directly from AudioDecoder
>     instead of being a subclass of AudioDecoderG722.
> 
>   - AudioDecoder now has a CngDecoderInstance member function, which
>     is implemented only by AudioDecoderCng. This replaces the previous
>     practice of calling AudioDecoder::state() and casting the result
>     to a CNG_dec_inst*. It still isn't pretty, but now the blemish is
>     plainly visible in the AudioDecoder class declaration.
> 
> R=henrik.lundin@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/24169005

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30879005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7629 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 00:45:58 +00:00
niklas.enbom@webrtc.org
8a232f65dd Revert 7625 "Don't use DCHECK when you need the side effects..."
Reverting since 7623 might depend on this one

> Don't use DCHECK when you need the side effects...
> 
> R=pbos@webrtc.org
> TBR=henrik.lundin@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/32369004

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7628 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 00:43:59 +00:00
buildbot@webrtc.org
795d003770 (Auto)update libjingle 79200114-> 79205306
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7627 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 00:14:02 +00:00
tkchin@webrtc.org
8125744a5f Cleanup RTCVideoRenderer interface.
RTCVideoRenderer should be a protocol not a class. This change includes
an adapter for use with the C++ apis. The video views have been refactored
to implement that protocol.

BUG=3795
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7626 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 23:06:15 +00:00
kwiberg@webrtc.org
b8425bc4f3 Don't use DCHECK when you need the side effects...
R=pbos@webrtc.org
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7625 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 22:10:18 +00:00
buildbot@webrtc.org
45ecf4c092 (Auto)update libjingle 79169148-> 79192489
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7624 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 21:48:54 +00:00
kwiberg@webrtc.org
9e525585fd Remove the state_ member from AudioDecoder
The subclasses that need a state pointer should declare them---with
the right type, not void*, to get rid of all those casts.

Two small but not quite trivial cleanups are included because they
blocked the state_ removal:

  - AudioDecoderG722Stereo now inherits directly from AudioDecoder
    instead of being a subclass of AudioDecoderG722.

  - AudioDecoder now has a CngDecoderInstance member function, which
    is implemented only by AudioDecoderCng. This replaces the previous
    practice of calling AudioDecoder::state() and casting the result
    to a CNG_dec_inst*. It still isn't pretty, but now the blemish is
    plainly visible in the AudioDecoder class declaration.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24169005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7623 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 21:18:47 +00:00
stefan@webrtc.org
7c29e8c2f3 Add support for VP9 in webrtc::Call and video_loopback.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7622 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 19:41:15 +00:00
stefan@webrtc.org
d839e0ab52 Reduce to 2 probes when probing for initial bandwidth.
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23359005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7621 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 19:33:55 +00:00
stefan@webrtc.org
db26247a9b Add UMA for measuring the diff between the BWE at 2 seconds compared to the BWE at 20 seconds when the BWE should have converged.
BUG=crbug/425925
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30819005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7620 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 19:32:10 +00:00
mcasas@webrtc.org
8944c9d08b AppRTCDemoActivity: use differnet Themes for different API levels
The current AndroidManifest.xml hardcodes a Theme that
is only available in Android L or later (Material). To
maintain backwards compatibility, and for better App
style, create a single Theme/Style and define it for
different APIs.

I tested this in two Nexus %, one with prerelease L
and another with a KK 4.4.2 and the Theme is indeed
automagically updated :)

Note that this is compatible with
https://webrtc-codereview.appspot.com/26979004/

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7619 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 17:26:22 +00:00
kjellander@webrtc.org
d367321a3f Add kjellander as PRESUBMIT.py OWNER
BUG=
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31949005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7618 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 17:06:31 +00:00
sprang@webrtc.org
dcebf2daa7 Reworked paced sender queue
Packet queue in the paced sender is now based on a priority queue rather than having a separate fifo-queue per priority level. This allows more flexible sorting and cleaner usage.

Packets with earlier capture times are now prioritized higher. In situations with high packet loss, the queue might contain packets from several subsequent frames. Retransmit packets from the earlier frames first, since the later ones will probably be dependent on these.

Also, don't force sending of packets after a certain time of inactivity or when packets grow too old, since this was causing consistent overuse on poor connections. Instead, drop frames in vie encoder if pacer queue is too long.

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

Review URL: https://webrtc-codereview.appspot.com/27869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7617 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 16:27:16 +00:00
pbos@webrtc.org
fad9aecbf5 Remove protected files from talk/PRESUBMIT.py.
All files may now be committed to.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/23359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7616 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 16:06:35 +00:00
pbos@webrtc.org
88ef632286 Falling back on single-stream on multiple SSRC.
Instead of failing, use one stream. Also clamp video min bitrate.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/31949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7615 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 15:29:29 +00:00
henrike@webrtc.org
28af64105b Presubmit was not whitelisting libjingle_tests.gyp or sound.gyp due to a missing comma leading to a concatenation of the two strings in the whitelist.
R=kjellander@webrtc.org
TBR=niklas.enbom@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/25999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7614 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 15:11:46 +00:00
stefan@webrtc.org
b3265accd9 Adds support for finch experiments to video_loopback.
Adds support for logging to stderr via -logs.

Enables abs-send-time by default.

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

Review URL: https://webrtc-codereview.appspot.com/23329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7613 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 14:57:14 +00:00
henrik.lundin@webrtc.org
52b42cb069 Fix problem with late packets in NetEq
Since r7255, it could happen that an old packet would block the decoding
process until enough packet was received for the buffer to flush. This
CL fixes that by:
- Partially reverting r7255;
- Remove recent old packets before taking a decision for GetAudio;
- Remove all old packets after a packet has been extracted for decoding;
- Adding tests for reordered packets.

BUG=chrome:423985
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7612 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 14:03:58 +00:00
pbos@webrtc.org
09cc686c8b Delete VideoReceiveStream channels in destructor.
R=stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/31909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 13:48:15 +00:00
kwiberg@webrtc.org
6de75ca3ed Remove the useless dummy state parameter to WebRtcPcm16b_DecodeW16
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7610 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 13:29:24 +00:00
kwiberg@webrtc.org
c78cf97ecb Remove the useless dummy state parameter to WebRtcG711_*
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7609 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 13:23:36 +00:00
perkj@webrtc.org
b5d045e94d ReAdd PeerConnectionInterface::AddStream to fix Chrome build.
AddStream(MediaStreamInterface* stream, const MediaConstraintsInterface* constraints);
This will be removed once Chrome has been updated.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7608 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 13:01:33 +00:00
tommi@webrtc.org
18de6f9622 Change the PeerConnection proxy templates to use blocking method calls instead of using Thread::Send.
The problem with Thread::Send is that it processes incoming pending messages and for the proxies,
this can mean that multiple incoming calls can concurrently run on the same thread, resulting in unexpected behavior.

See e.g. crbug.com/429740 (and more)

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7607 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 12:08:48 +00:00
kwiberg@webrtc.org
721ef633d0 Remove the codec_type_ member from AudioDecoder
It isn't actually required, as evidenced by the comparative ease with
which it can be removed.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7606 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 11:51:46 +00:00
perkj@webrtc.org
c2dd5ee2c0 Prepare for removal of PeerConnectionObserver::OnError.
Prepare for removal of constraints to PeerConnection::AddStream.

OnError has never been implemented and has been removed from the spec.
Also, constraints to PeerConnection::AddStream has also been removed from the spec and have never been implemented.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7605 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 11:31:29 +00:00
stefan@webrtc.org
f37145f685 Enables AIMD control by default.
BUG=1788
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7604 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 09:08:21 +00:00
henrik.lundin@webrtc.org
b0f4b3da05 Improving error message from neteq_rtpplay
If a packet with unknown RTP payload type is inserted, this CL
will make sure that the error message is a little more detailed
and gives a better understadning of what to do.

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

Review URL: https://webrtc-codereview.appspot.com/27909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7603 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 08:53:10 +00:00
buildbot@webrtc.org
a663d90ae3 (Auto)update libjingle 79104430-> 79104922
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7602 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 22:29:18 +00:00
glaznev@webrtc.org
5f38c8d1b8 Android AppRTCDemo improvements:
- Add a room list to ConnectActivity with buttons to add/remove rooms.
- Add loopback call button.
- Add option to toggle full screen / letterbox video.
- Add camera fps settings.
- Fix device to landscape orientation for HD video until issue 3936
will be fixed.
- Fix a few crashes by avoiding calling peer connection and
GAE signaling function while connection is closing.
- Better handling GAE channel error - catch channel exceptions and
display dialog with error messages.

BUG=3939, 3935
R=kjellander@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7601 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 22:18:52 +00:00
andrew@webrtc.org
5804936052 Add format members to AudioConverter for DCHECKing.
And use a std::min. Post-commit fixes after:
https://review.webrtc.org/30779004/

TBR=kwiberg

Review URL: https://webrtc-codereview.appspot.com/25059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7600 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 21:32:14 +00:00
marpan@webrtc.org
e451b756a8 Update rate control parameter in vp9 test.
TBR=hellner@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7599 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 21:26:08 +00:00
marpan@webrtc.org
4765ca55f9 Roll chromium_revision: 28d1981..d3db2ff
Pick up the libvpx roll: https://codereview.chromium.org/674753002

Summary of changes (28d1981..d3db2ff/DEPS):
* third_party/android_tools 36bf7ac..ea50ccc
* third_party/boringssl 7ea8481..751e889
* third_party/icu 8ac906f..d8b2a9d
* third_party/libvpx efe9712..2e5ced5
* third_party/usrsctp/usrsctplib
* tools/gyp 1990:1991
* tools/swarming_client a57d7db..bcb3bc3

Clang is not updated in this roll.

Made the change getchar() --> getc(stdin) as seems like getchar() isn't supported on android anymore.
(getchar() was causing the error: undefined reference to '__srget')

Update rate control parameter in vp9 test.

R=andrew@webrtc.org
TBR=ajm@google.com

Review URL: https://webrtc-codereview.appspot.com/23229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7598 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 20:10:26 +00:00
andrew@webrtc.org
f866b2d9f9 Restore the void return type on WriteWavHeader.
Karl pointed out that the user can check the validity of the input
parameters with CheckWavParameters prior to calling.

TBR=kwiberg

Review URL: https://webrtc-codereview.appspot.com/23339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7597 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 18:20:06 +00:00
andrew@webrtc.org
b81e304ac0 replace inline assembly WebRtcNsx_AnalysisUpdate by intrinsics.
The modification only uses the unique part of the analysis_update
 function. Pass byte to byte conformance test on both ARMv7 and AArch64,
 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                   |    15.61% |    20.15% |     14.89% |
| Neon inline asm (LLVM 3.4) |    25.98% |    33.96% |     18.18% |
| Neon intrinsics (GCC 4.6)  |    22.06% |    27.01% |     19.24% |
| Neon intrinsics (GCC 4.8)  |    17.28% |    18.23% |     18.55% |
| Neon intrinsics (LLVM 3.4) |    21.02% |    19.98% |     16.76% |

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28849004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7596 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 17:17:51 +00:00
henrik.lundin@webrtc.org
f9471807a2 Add Opus support to neteq_rtpplay
BUG=2416
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7595 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 15:19:58 +00:00
pbos@webrtc.org
96a93259b3 Implement external decoder support in WebRtcVideoEngine2.
R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/30839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7594 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 14:46:44 +00:00
stefan@webrtc.org
548b228c91 Add UMA metrics for the initial (after two seconds) packet loss, round-trip time and bandwidth estimate of a WebRTC call.
BUG=crbug/425925
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7593 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 14:42:43 +00:00
asapersson@webrtc.org
96dc685143 Add stats for video:
- number of sent/received RTCP NACK/FIR/PLI per minute
- percentage of unique sent/received NACK requests
- percentage of discarded/duplicated packets by the jitter buffer
- permille of sent/received key frames

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

Review URL: https://webrtc-codereview.appspot.com/24959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7592 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 14:40:38 +00:00
henrik.lundin@webrtc.org
2236267b5e Disable PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate under MSan
This test is flaky on MSan bots.

BUG=3980
TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7591 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 13:38:50 +00:00
braveyao@webrtc.org
bf09976e86 Add more sanity checks to workaround the unidentified problem that CaptureThread is still running while related resouces are destroyed already.
BUG=
TEST=auto test
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7590 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 09:58:40 +00:00