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Author SHA1 Message Date
kjellander@webrtc.org
cc483b7379 Roll chromium_revision 601e6f3..b0c3ed3 (315263:316737) 
Moved LSan suppressions from tools/lsan/suppressions.txt to compiled-in
suppressions similar to the Chromium changes in
https://codereview.chromium.org/924923002
I will remove tools/lsan after committing this and the bots are updated to
not specify it.

Other relevant changes:
* src/buildtools: da0df3f..5c5e924
* src/third_party/android_tools: f6e2370..fd5a8ec
* src/third_party/boringssl/src: 8f5e2eb..d306f16
* src/third_party/openmax_dl: 81318c1..21c8abe
Details: 601e6f3..b0c3ed3/DEPS

Clang version was not updated in this roll.

R=glider@chromium.org, henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37299004

Cr-Commit-Position: refs/heads/master@{#8406}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8406 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 10:38:11 +00:00
pbos@webrtc.org
b4987bfc24 Send black frame with previous size when muting. 
Instead of sending a black frame that's the size of the VideoFormat send
a black frame in the format we're already sending. This prevents
expensive encoder reconfiguration when the sending format is a different
resolution. This speeds up setting a null capturer (removing the
capturer) significantly as it doesn't entail an encoder reconfiguration.

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

Review URL: https://webrtc-codereview.appspot.com/39179004

Cr-Commit-Position: refs/heads/master@{#8405}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8405 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 10:13:45 +00:00
minyue@webrtc.org
7d721eea14 Adding speech_expand_rate to NetEQ Network Statistics. 
There have been requests for separating rate of expanded speech samples from noise samples.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37309004

Cr-Commit-Position: refs/heads/master@{#8404}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8404 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 10:02:20 +00:00
magjed@webrtc.org
3864363e2c cricket::VideoFrame: Refactor CopyToBuffer into base class 
It’s possible to implement cricket::VideoFrame::CopyToBuffer using the virtual interface. This removes the need for subclasses to implement their own versions. This CL also fixes a bug in cricket::VideoFrame::CopyToPlanes which currently assumes that GetUPitch() == GetVPitch(), otherwise it may segfault.

I think this CL should land regardless, but the main purpose is to pave the way for for planned changes to I420VideoFrame. See https://review.webrtc.org/38879004.

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

Review URL: https://webrtc-codereview.appspot.com/39889004

Cr-Commit-Position: refs/heads/master@{#8403}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8403 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 09:19:45 +00:00
magjed@webrtc.org
dd4a8da68a Remove DISABLE_YUV flag 
R=fbarchard@google.com, pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41979004

Cr-Commit-Position: refs/heads/master@{#8402}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8402 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 08:47:43 +00:00
jansson@webrtc.org
97aaf68fed Bump to version 42. 
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40739004

Cr-Commit-Position: refs/heads/master@{#8401}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 08:20:23 +00:00
decurtis@webrtc.org
bfa3c7253f Don't call g_thread_init on glib >=2.31.0 
g_thread_init() is deprecated in glib 2.31.0 and later. This will call
g_thread_ini() only when compiling against older versions of glib.

BUG=1971,chromium:253566
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40019004

Cr-Commit-Position: refs/heads/master@{#8400}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8400 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 21:23:13 +00:00
pkasting@chromium.org
e9facf8bb3 Add range checks in a variety of places where the values will subsequently be 
expected to be 0-127.

BUG=none
TEST=none
R=juberti@webrtc.org
TBR=henrika

Review URL: https://webrtc-codereview.appspot.com/37759004

Cr-Commit-Position: refs/heads/master@{#8399}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8399 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 20:37:35 +00:00
aluebs@webrtc.org
27669f320b Apply good settings to Beamformer 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33219004

Cr-Commit-Position: refs/heads/master@{#8398}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8398 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 19:24:37 +00:00
guoweis@webrtc.org
b08f4045ec Fix issue 4061. 
Issue was that the longest 0 detection wasn't done when there is only one 0 octet. The purpose of this detection is to make sure we don't also compression 0 octet sequences which are not longest.

BUG=4061
R=juberti@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33309004

Cr-Commit-Position: refs/heads/master@{#8397}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8397 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 19:01:14 +00:00
mflodman@webrtc.org
0abc6011b9 Remove SetCaptureDelay from the RTP module. 
This is a small step in getting rid of the default module, but also to
eventually delete FrameProviderBase completely.

BUG=769
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34229004

Cr-Commit-Position: refs/heads/master@{#8396}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8396 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 16:36:48 +00:00
stefan@webrtc.org
7663684258 Implement the Nada rmcat proposal within the simulation framework. 
This first CL focuses only on the bandwidth estimation parts of NADA, and doesn't contain the rate smoothing. It is still missing slow start functionality.

https://datatracker.ietf.org/doc/draft-zhu-rmcat-nada/

BUG=
R=sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35219004

Cr-Commit-Position: refs/heads/master@{#8395}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8395 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 16:04:17 +00:00
jmarusic@webrtc.org
71b35a4ce4 iLBC: Use uint8_t[] for byte arrays 
BUG=909

This is the same as https://review.webrtc.org/41779004/ with the review comments addressed.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40769004

Cr-Commit-Position: refs/heads/master@{#8394}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8394 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 16:02:46 +00:00
magjed@webrtc.org
640313ce4f WebRtcVideoCapturer: Remove dead code |OnIncomingCapturedEncodedFrame| 
The end goal except cleanup is to remove webrtc::VideoFrame.

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

Review URL: https://webrtc-codereview.appspot.com/36079004

Cr-Commit-Position: refs/heads/master@{#8393}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8393 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 15:10:41 +00:00
magjed@webrtc.org
7a91acb94a ViECapturer: Remove unimplemented function declaration |DeliverCodedFrame| 
The end goal except cleanup is to remove webrtc::VideoFrame.

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

Review URL: https://webrtc-codereview.appspot.com/35129004

Cr-Commit-Position: refs/heads/master@{#8392}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8392 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 14:57:22 +00:00
perkj@webrtc.org
1a38a51119 Add default implementation to VideoSourceInterface of Stop and Restart. 
This is to make sure Chrome does not break when rolling. This should be reverted once
Chrome has been updated.

Please see:
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac/builds/16556/steps/compile/logs/stdio

BUG=4303
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35229004

Cr-Commit-Position: refs/heads/master@{#8391}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8391 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 14:51:43 +00:00
pbos@webrtc.org
a28a91d2f0 Fix data race for RTCPReceiver stats callback. 
Annotates the callback which identifies the bug, then fixes it.

R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/40009004

Cr-Commit-Position: refs/heads/master@{#8390}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 14:45:44 +00:00
perkj@webrtc.org
8f605e8911 Add VideoSource::Stop and Restart methods. 
The purpose is to make sure that start and stop is called on the correct thread on Android. It also cleans up the Java VideoSource implementation.

BUG=4303
R=glaznev@webrtc.org, magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39989004

Cr-Commit-Position: refs/heads/master@{#8389}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8389 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 13:54:42 +00:00
magjed@webrtc.org
959dac7498 VideoCaptureImpl: Remove unused member variable |_capture_encoded_frame| 
The end goal except cleanup is to remove webrtc::VideoFrame.

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

Review URL: https://webrtc-codereview.appspot.com/37219004

Cr-Commit-Position: refs/heads/master@{#8388}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8388 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 13:44:46 +00:00
pbos@webrtc.org
4dd40d6b88 Signal threads for faster receiver destruction. 
Unblocks pending threads (render thread + decoder thread) when
destroying renderers and shutting down decoders.

Speeds up SetLocalDescription significantly (10x or so) under
WebRtcVideoEngine2 but also shutdown times in ~ViEChannel and
~ViEReceiver in general.

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

Review URL: https://webrtc-codereview.appspot.com/41959004

Cr-Commit-Position: refs/heads/master@{#8387}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 13:23:27 +00:00
mflodman@webrtc.org
0a7d4eed98 Remove usage of BitrateController in VoiceEngine. 
Bitrate controller is used in VoiceEngine to smoothen the fraction loss
from RTCP report blocks. This CL removes the usage of the
BitrateController and calculates its own fraction loss average insted.
This introduces some duplicated code between BitrateController and
Channel, but removes processing overhead and the incorrect bandwidth
estimation numbers reported by the bitrate controller.

BUG=4310
TEST=voe_cmd_test with network simulator
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39999004

Cr-Commit-Position: refs/heads/master@{#8386}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8386 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 12:57:44 +00:00
minyue@webrtc.org
f9b5c1b3d0 Removing CELT. 
CELT is not supported in WebRTC/Libjingle. There are a few left-over in our code base. They are cleaned up in this CL.

BUG=
R=pbos@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36099004

Cr-Commit-Position: refs/heads/master@{#8385}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 12:37:14 +00:00
minyue@webrtc.org
2c1bcf2cb4 Adding decoded_fec_rate to NetEQ Network Statistics. 
A statistic is introduced to reflect the actual benefits of Opus FEC. It shows what percentage of samples in the rendered audio come from FEC data.

BUG=3867
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34969004

Cr-Commit-Position: refs/heads/master@{#8384}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8384 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 10:17:48 +00:00
mflodman@webrtc.org
290cb56dca Remove TimeToSendPacket and TimeToSendPadding from the default module. 
Thie CL moves the default RTP module logic for TimeToSendPacket and
TimeToSendPadding to PayloadRouter class and asserts on usage of the
default module.

BUG=769
TEST=New unittest.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33319004

Cr-Commit-Position: refs/heads/master@{#8383}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 10:15:47 +00:00
kjellander@webrtc.org
c0fc4dd87c Add 'mac_x64' trybot to default set. 
This trybot was recently added since we've run into
some failures only happening for Mac 64 Debug.

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41969004

Cr-Commit-Position: refs/heads/master@{#8382}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8382 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 08:30:41 +00:00
pbos@webrtc.org
86196c4f48 Setup encoders inexpensively before first frame. 
Modifies WebRtcVideoSendStream to use a default width/height of 16px.
This significantly reduces SetRemoteDescription time under
WebRtcVideoEngine2. Also preventing (expensive) reconfigurations due to
incoming frames when the channel is not sending yet.

Tests have been modified to generate a frame before expecting a certain
encoder size to have been configured.

Also adding tracing to WebRtcVideoSendStream::InputFrame as it can lead
to reconfigurations of the encoder which is expensive and it should show
up in chrome://tracing.

BUG=1788
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42369004

Cr-Commit-Position: refs/heads/master@{#8381}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8381 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 21:02:20 +00:00
henrik.lundin@webrtc.org
34509d9f33 Fix an issue with comfort noise in ACMGenericCodecWrapper 
In some cases it was not possible to set another payload type for CNG
than the default one. This CL fixes this. The problem was also
dependent on whether the comfort noise codec was registered before or
after the speech codec.

A test is implement to expose the bug, registering comfort noise at a
non-default payload type, and both before and after the speech codec.

BUG=4228
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35199004

Cr-Commit-Position: refs/heads/master@{#8380}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8380 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 16:02:40 +00:00
stefan@webrtc.org
e9f0f591b5 Enable bitrate probing by default in PacedSender. 
BUG=crbug:425925
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33359004

Cr-Commit-Position: refs/heads/master@{#8379}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8379 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 15:48:29 +00:00
henrik.lundin@webrtc.org
fbc347f2ef Re-land r8342 "Switch to using AudioEncoderIsac instead of ACMISAC"" 
This reverts r8372, with a bug fix: allowing zero rate in
AudioEncoderIsac::Config. Without this fix, setting the rate to zero
triggered a CHECK. Existing callers assumed that zero was a valid
value. Setting the rate to zero will result in the default rate 32000
being set.

BUG=4228,chromium:458638
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org
TBR=tina.legrand@webrtc.org
CC=jmarusic@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39159004

Cr-Commit-Position: refs/heads/master@{#8378}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8378 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 14:28:45 +00:00
kjellander@webrtc.org
ce22f13f0e GN: Changes for vp9, opus and direct trace 
Corresponding GN changes for
https://webrtc-codereview.appspot.com/34099004/

BUG=4185
R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/40669004

Cr-Commit-Position: refs/heads/master@{#8377}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8377 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 12:47:45 +00:00
kjellander@webrtc.org
e35fa96cbe Move isacfix.gypi and isac.gypi 
Move webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.gypi
and webrtc/modules/audio_coding/codecs/isac/main/source/isac.gypi to
webrtc/modules/audio_coding/codecs/isac to pass recently
added _CheckNoSourcesAboveGyp presubmit rule.

BUG=4002
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37269004

Cr-Commit-Position: refs/heads/master@{#8376}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8376 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 12:47:22 +00:00
sprang@webrtc.org
0200f70792 Set webrtc_rtp category to be disabled by default. 
Should not affect webrtc standalone. For chromium, disabling helps
mitigate viewing performance problems.

BUG=chromium:441440
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41909004

Cr-Commit-Position: refs/heads/master@{#8375}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8375 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 12:06:48 +00:00
stefan@webrtc.org
14b0279416 Break out code from bloated files in the BWE simulator. 
No changes to functionality.

BUG=4173
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34209004

Cr-Commit-Position: refs/heads/master@{#8374}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8374 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 12:03:19 +00:00
kjellander@webrtc.org
0f7f161ed6 Add audio_coding module OWNERS file. 
It should simplify things to have an
OWNERS file at the top level of audio_coding, in addition
to the lower ones.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39149004

Cr-Commit-Position: refs/heads/master@{#8373}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 09:53:47 +00:00
henrik.lundin@webrtc.org
4dc0003bed Revert r8342 "Switch to using AudioEncoderIsac instead of ACMISAC" 
BUG=chromium:458638
TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33349004

Cr-Commit-Position: refs/heads/master@{#8372}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-14 09:42:41 +00:00
aluebs@webrtc.org
30142bbe07 Add arraysize to overrides to avoid macros redefinitions in Chromium 
This fixes the Webrtc in Chrome FYI bots breaking.

TBR=ajm

Review URL: https://webrtc-codereview.appspot.com/36149004

Cr-Commit-Position: refs/heads/master@{#8371}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8371 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-14 02:46:13 +00:00
guoweis@webrtc.org
d3b453be17 Remove the incremental IP address behavior from virtualsocketserver 
VirtualSocketServer, when binding to any address (all 0s), will assign a unique IP address by incrementing the IP address, resulted in 0.0.0.1. However, this breaks the testing of 4276 where we bind to all 0s and expect the local address should remain all 0s.

BUG=4276
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35189004

Cr-Commit-Position: refs/heads/master@{#8370}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8370 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-14 00:44:20 +00:00
pthatcher@webrtc.org
3341b401cc Fix bug parsing media descriptions: the final field isn't a codec type for any of DTLS/SCTP, SCTP, or SCTP/DTLS. 
BUG=none
TEST=none
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34029004

Cr-Commit-Position: refs/heads/master@{#8369}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8369 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 21:14:44 +00:00
aluebs@webrtc.org
92a19bcbd7 Simplify mask calculation 
There are only 2 things that prevent the output to be bit-exact:
* The zero initialization of the postfilter_mask_ and high_pass_postfilter_mask_, which only afects the first blocks.
* The re-tuning of the target presence estimation, since only the bins between low_average_start_bin_ and high_average_end_bin_ are of interest.
This latter was not taken into account before.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35139004

Cr-Commit-Position: refs/heads/master@{#8368}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8368 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 19:38:22 +00:00
stefan@webrtc.org
56cb0ea99c Add support for bi-directional simulations by having both an uplink and a downlink. 
Partially make PacketReceiver a source which should be possible to hook up to the network.

Require that flow ids are set in all constructors.

BUG=4173
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36059004

Cr-Commit-Position: refs/heads/master@{#8367}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8367 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 15:46:52 +00:00
pbos@webrtc.org
d5ce2e63df Remove EventWrapper::Reset(). 
This simplifies the event wrapper which we've recently found issues in.
Also refactoring EndToEndTest.RespectsNetworkState to not depend on it.

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

Review URL: https://webrtc-codereview.appspot.com/41939004

Cr-Commit-Position: refs/heads/master@{#8366}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 14:58:38 +00:00
guoweis@webrtc.org
5a7dc39277 This is a code clean up. No functional change intended. 
Consolidate the enum for capturer/frame rotation we use through out the code base.

BUG=4145
R=mflodman@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39859004

Cr-Commit-Position: refs/heads/master@{#8365}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 14:32:13 +00:00
minyue@webrtc.org
a8cc3440b1 Allowing RED decoding for Opus. 
BUG=4247
TEST=reproduced and fixed the bug
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41809004

Cr-Commit-Position: refs/heads/master@{#8364}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8364 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 14:02:17 +00:00
perkj@webrtc.org
96e4db9bea Split peerconnection_jni.cc into separate files. 
For now:
java_helpers - JNI convenience functions etc. Can in theory be moved to libjingle / webrtc general one day.
classreferenceholder - app/webrtc specific Java class loader.
androidvideocapturer_jni - the jni part of the video capturer I added.
peerconnection_jni - all the rest.

This also move all jni specifics into ns webrtc_jni to avoid naming collision.

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38099004

Cr-Commit-Position: refs/heads/master@{#8363}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 12:47:21 +00:00
solenberg@webrtc.org
8db5854eb0 Fix potential flakiness in voe_auto_test. 
BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41929004

Cr-Commit-Position: refs/heads/master@{#8362}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8362 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 12:19:42 +00:00
solenberg@webrtc.org
2adf4c4edd Re-enable BWE tests using baseline files. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39119004

Cr-Commit-Position: refs/heads/master@{#8361}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8361 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 12:05:16 +00:00
henrika@webrtc.org
58f6f01acc WebRTC now compiles for enable_android_opensl=1. 
Default is enable_android_opensl=0 but we should build for OpenSL as well.

BUG=4293
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40719004

Cr-Commit-Position: refs/heads/master@{#8360}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8360 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 11:36:12 +00:00
solenberg@webrtc.org
40fdb8ab96 Remove flaky test cases from peerconnection_unittest. The chain of API calls is tested from top to bottom anyway. 
BUG=3871
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41879004

Cr-Commit-Position: refs/heads/master@{#8359}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8359 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 11:09:43 +00:00
bjornv@webrtc.org
ba97ea69f0 audio_coding/codec/ilbc: Removed usage of macro WEBRTC_SPL_MUL_16_16 
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)

The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)

Some other minor code cleanup also exists.

BUG=3348, 3353
TESTED=locally on Mac and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34179004

Cr-Commit-Position: refs/heads/master@{#8358}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8358 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 09:52:42 +00:00
mflodman@webrtc.org
2bd299a172 Remove call to RtpRtcp::RegisterSendPayload for the default RTP module. 
The send payload type is only checked in RTPSender::CheckPayloadType,
which in turn is only called from SendOutgoingData and never from the
default module anylonger.

BUG=769
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39949004

Cr-Commit-Position: refs/heads/master@{#8357}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 09:52:17 +00:00