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Author SHA1 Message Date
brettw@chromium.org
0867f69cc6 Convert GN visibility to be lists. 
This is a followup to my previous patch that missed this case.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7137 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 16:24:11 +00:00
andresp@webrtc.org
33aa095896 Simplify gyp rules on video_render_module. 
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 14:48:48 +00:00
houssainy@google.com
e0761d06b0 Fix printing of error stack in rtcbot when a test fails via test.fail(). 
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7134 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 14:35:35 +00:00
kjellander@webrtc.org
49fa212bcd Fix compile error on JDK 1.7. 
JDK 1.7 gives an error like this:
warning: [static] static method should be qualified by type name

R=pbos@webrtc.org
TBR=henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/29399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7133 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 12:35:59 +00:00
henrik.lundin@webrtc.org
23a5e3c3b0 Remove DestructEncoderInst and its codec-specific implementations. 
This method is seemingly never called.

BUG=none
TEST=none
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7131 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 08:52:26 +00:00
andrew@webrtc.org
4ca66d691e include cstdlib for free() and abort() 
This previous CL added uses of free() and abort() without including cstdlib:
https://webrtc-codereview.appspot.com/22449004

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23559004

Patch from Mostyn Bramley-Moore <mostynb@opera.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7127 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 03:24:36 +00:00
guoweis@webrtc.org
fa603981f2 Add a new class InterfaceAddress inherited from IPAddress to keep track of IPv6 Address flags. 
Skeleton put in place in Network::GetFilterIPs() which will be used to
filter addresses

BUG=3773
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7126 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 23:42:40 +00:00
brettw@chromium.org
87ff9c8efa Fix up configs applying to GN build. 
The audio_processing target didn't have the build configs applying to it which led to some logging errors.

TBR=kjellander

Review URL: https://webrtc-codereview.appspot.com/22339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7125 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 23:34:56 +00:00
fbarchard@google.com
a941970d4a Change explicit static cast from int to uint16_t to implicit cast of 0u. 
BUG=3663
TESTED=local windows build with VS2013.
R=harryjin@google.com, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7123 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 21:37:27 +00:00
brettw@chromium.org
9fe11010f7 Fix the RTC+Chromium GN build. 
LOGGING_INSIDE_WEBRTC was being set in the inherited config, whereas in the GYP build this define is not inherited. This caused duplicate logging macros to be defined in Chrome files dependening on WebRTC targets.

Move LOGGING_INSIDE_WEBRTC to the common config (non-inherited).

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7122 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 19:15:33 +00:00
jiayl@webrtc.org
22406fcc9b TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH. 
BUG=3570
R=juberti@webrtc.org, mallinath@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7070

Review URL: https://webrtc-codereview.appspot.com/20999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7120 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 15:44:05 +00:00
houssainy@google.com
04b853b56a Bot Browser files moved to /bot/browser/ 
because android files will be a different and will need to add more files for Android.

There was a CL to move the browser files form bot/browser/ to /bot/ as i thought it will be the same files used for Android.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7119 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 14:50:09 +00:00
guoweis@webrtc.org
4bbd3c83a8 fix a bug in the logic when new Networks are merged. This happens when 
we have 2 networks with the same key

BUG=410554 in chromium

http://code.google.com/p/chromium/issues/detail?id=410554

Corresponding change in chromium is
https://codereview.chromium.org/536133003/

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19249005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7117 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 13:54:45 +00:00
pbos@webrtc.org
b420191743 Expose VideoEncoders with webrtc/video_encoder.h. 
Exposes VideoEncoders as part of the public API and provides a factory
method for creating them.

BUG=3070
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7114 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 10:40:56 +00:00
andrew@webrtc.org
641bda6f9c Initialize ChannelBuffer's memory to avoid uninitialized reads. 
Removed the zero out memset in this change:
https://review.webrtc.org/24469004/

assuming it was unneeded. Dr. Memory taught me that assupmtion was
invalid. linux_memcheck try runs might have caught this, if they
weren't flaking out on unrelated stuff.

TBR=claguna@google.com

Review URL: https://webrtc-codereview.appspot.com/28429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7113 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 23:11:44 +00:00
brettw@chromium.org
519c9e207d Convert GN visibility to be a list. 
GN visibility currently allows either string or list types, but this is causing
some problems for some templates. I'm going to require it to be lists, so am
changing all callers before pushing the new binary.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7111 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 22:45:18 +00:00
andrew@webrtc.org
17454f79dc Add ctors to ChannelBuffer to enable copying on construction. 
Also:
- Fix the constness of some parameters.
- Add more const overloads.
- Use DCHECK in place of assert.
- Removed an unnecessary memset.

R=claguna@google.com

Review URL: https://webrtc-codereview.appspot.com/24469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 20:27:04 +00:00
henrik.lundin@webrtc.org
c64246f42c Set a default speech type in iSAC wrapper 
If the decoder encounters an error, it may leave the speech type
unassigned, leading to a use-of-uninitialized-value in subsequent lines.

BUG=crbug/411162
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7104 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 13:40:58 +00:00
henrik.lundin@webrtc.org
ed8bcd3ac5 Starting to implement the new ACM API 
The new implementation class is called AudioCodingImpl, and will in the
end replace AudioCodingModuleImpl.

This is work in progress.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7103 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 13:13:19 +00:00
houssainy@google.com
9600519147 Adding the ability to test on Chrome for Android. 
use "android-chrome" as type in rtcbot running command.
Example: node test.js android-chrome

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 13:01:40 +00:00
bjornv@webrtc.org
37c39f3784 audio_processing: Removed use of macro WEBRTC_SPL_UMUL_16_16 
The macro replaced is a trivial multiplication after explicit casts to uint16_t and uint32_t. This CL replaces its use with "*" and adds explicit casts if necessary.

Affected components:
* AECMobile
* AGC
* Noise Suppression (fixed point version)

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

Review URL: https://webrtc-codereview.appspot.com/27389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7101 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 11:21:56 +00:00
bjornv@webrtc.org
0d394f3609 video_processing: Removed usage of WEBRTC_SPL_UMUL_16_16 
The trivial macro WEBRTC_SPL_UMUL_16_16 is nothing but plain mutliplication of casted values. This CL explicitly use "*" at place and casts if necessary.

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

Review URL: https://webrtc-codereview.appspot.com/23499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7100 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 11:19:39 +00:00
houssainy@google.com
c77e4d6aef - Adding AndroidDeviceManager to botManager.js to help in selecting devices, in case running test on Android devices. 
- Select BotType using nodeJs terminal command.

- ping_pong.js test added.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7099 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 10:36:11 +00:00
stefan@webrtc.org
fe16167507 Fix RTT calculations for send-only channels. 
As we don't know the SSRC of the other end in a send-only channel since we haven't received packets from that end, we are required to assume that the SSRC of the first report block is the correct SSRC to use for RTT calculations.

BUG=3781
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7097 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 08:45:25 +00:00
sprang@webrtc.org
c30e9e2300 Ignore FEC packet in stats, if it is first packet on ssrc. 
BUG=chrome:410456
R=mflodman@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7096 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 08:20:18 +00:00
kjellander@webrtc.org
6d08ca6379 GN: Prefix WebRTC specific variables with "rtc_" 
BUG=3441
TESTED=Trybots + Running GN in a Chromium checkout with
src/third_party/webrtc symlinked to the WebRTC checkout
with this CL applied, both with the default GN settings
and using: --args="os=\"android\" cpu_arch=\"arm\""

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/27379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-07 17:36:10 +00:00
kjellander@webrtc.org
f68cf93e1b Add video_capture_tests_apk_target 
In https://codereview.chromium.org/500423004/ the
target that was previously used to build the Android APK
tests was removed. When building these tests from a
standalone checkout, the video_capture_tests_apk target
was missing in the chain of targets that gets generated
into the 'all' target.

BUG=3764
TESTED=Trybots.
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7094 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-07 17:35:51 +00:00
kjellander@webrtc.org
a781f68712 Fix rm command for class cleanup in r7091 
In https://webrtc-codereview.appspot.com/20339004
the rm command was missing 'r' for recursive mode.

TBR=henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/26379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7092 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-06 22:11:28 +00:00
kjellander@webrtc.org
9510022e1f Cleanup temporary class files for OpenSlDemo 
I've seen tryjobs failing when they shouldn't on
the Android trybots and I suspect this might have
something to do with it.

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7091 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-06 18:03:45 +00:00
henrik.lundin@webrtc.org
8f073c5054 Create a new interface for AudioCodingModule 
This is a first draft of the interface, and is work in progress.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7085 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 13:16:23 +00:00
kjellander@webrtc.org
fed47dc205 Drop buildbot_tests.py script 
This is no longer used since the buildbots have moved
over to recipes (where these arguments are configured).
See https://code.google.com/p/chromium/codesearch#chromium/tools/build/scripts/slave/recipe_modules/webrtc/api.py&l=73
for details.

This is essentially a revert of
https://webrtc-codereview.appspot.com/1021006

BUG=None
TESTED=None
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7079 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 08:25:38 +00:00
henrik.lundin@webrtc.org
bcf75e32a3 Modifying audio_coding/codecs/OWNERS 
Adding myself.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7077 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 07:18:50 +00:00
bjornv@webrtc.org
c2c4117477 common_audio: Replaced WEBRTC_SPL_LSHIFT_U32 with << in audio_processing 
Affected components:
* AECMobile
  - Added a help function since the same operation was performed several times.
* Auto Gain Control
* Noise Suppression (fixed point)

BUG=3348,3353
TESTED=locally on Linux
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7076 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 06:01:53 +00:00
henrike@webrtc.org
f1427c6731 Revert 7070 "TurnPort should retry allocation with a new address on error 
STUN_ERROR_ALLOCATION_MISMATCH."

TBR=jiayl@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/15359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7072 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 22:21:33 +00:00
jiayl@webrtc.org
574f2f60fe TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH. 
BUG=3570
R=juberti@webrtc.org, mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7070 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 19:11:34 +00:00
aluebs@webrtc.org
021e76fd39 Add support for WAV output in audioproc 
The default output is a WAV file, except if the --pcm_output flag is set.

BUG=webrtc:3359
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 18:12:00 +00:00
brettw@chromium.org
afa77cd803 Add direct_dependent_config to desktop_capture in GN build. 
This allows us to remove some configs in the Chrome build that should come
automatically from depending on this target.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7067 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 17:00:55 +00:00
andresp@webrtc.org
fa822b940f Fix strange owners files with comments that crashs "git cl presubmit" 
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7064 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 14:25:07 +00:00
kjellander@webrtc.org
79ee97cf43 [MIPS] Fix gn gen failure for MIPS in webrtc 
Fixes the following failure for mips:
"ERROR at //third_party/webrtc/BUILD.gn:136:7: Undefined variable for +=.
      cflags += [ "-mhard-float" ]
      ^-----
I don't have something with this name in scope now."

BUG=3441
TEST=In Chromium. Passing compile locally on Linux using:
gn gen out-gn/mips --args="is_debug=false os=\"android\" cpu_arch=\"mipsel\"" --verbose &&  ninja -C out-gn/mips all
gn gen out-gn/arm --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" --verbose &&  ninja -C out-gn/arm all
gn gen out-gn/x86-linux --args="is_debug=false os=\"linux\"" --verbose &&  ninja -C out-gn/x86-linux webrtc

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15349004

Patch from Gordana Cmiljanovic <Gordana.Cmiljanovic@imgtec.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7063 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 14:10:49 +00:00
houssainy@google.com
38ef664418 Moving the api.js and bot.js to /rtcbot/bot/ to be shared between 
/borwser and /android

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7062 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 13:44:47 +00:00
andresp@webrtc.org
262e676a08 Reland rev 7041 with BUILD.gn files. 
Original description:
  Audio codecs to include webrtc/typedefs.h

  Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h

  CL Generated with:
  $ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g"

BUG=3777
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 13:28:48 +00:00
bjornv@webrtc.org
3cbd6c26c8 Fix MSVC warnings about value truncations, webrtc/common_audio/ edition. 
This changes some method signatures to better reflect how callers are actually
using them.  This also has the tendency to make signatures more consistent about
e.g. using int (instead of int16_t) for lengths of things like vectors, and
using int16_t (instead of int) for e.g. counts of bits in a value.

This also removes a couple of functions that were only called in unittests.

BUG=3353,chromium:81439
TEST=none
R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7060 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 13:21:44 +00:00
henrik.lundin@webrtc.org
f6ab6f86e7 Rename Audio[Multi]Vector.CopyFrom to .CopyTo 
The name of the copy method was confusing. This change makes the
code easier to read where the method is used.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7059 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 10:58:43 +00:00
kjellander@webrtc.org
3c0aae17f0 Change gflags and gmock includes to be full paths. 
This will fix PRESUBMIT warnings developers will get due to
r7014 and r7020.

Also some minor style cleanup in:
webrtc/modules/audio_coding/main/test/RTPFile.cc
webrtc/modules/audio_coding/neteq/test/RTPjitter.cc

BUG=
R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 09:55:40 +00:00
kwiberg@webrtc.org
51bb33cc18 ACMOpus: Remove useless member variable fec_enabled_ 
R=henrik.lundin@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7057 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 08:42:44 +00:00
henrik.lundin@webrtc.org
7825b1abf9 Add support for multi-channel DTMF tone generation 
This CL opens up support for DTMF tones to be played to multi-channel
outputs. The same tones are replicated across all channels. Unit tests
are updated.

Also adding a new method AudioMultiVector::CopyChannel.

BUG=crbug/407114
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7056 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 07:39:21 +00:00
asapersson@webrtc.org
9d453931c5 Change return value for number of discarded packets to be int. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 07:07:44 +00:00
stefan@webrtc.org
01581da711 Fix audio/video sync when FEC is enabled. 
Also improves the tests by adding a test case for FEC, and running the a/v sync
tests with NACK and simulated packet loss.

BUG=crbug/374104
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7053 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 06:48:14 +00:00
andresp@webrtc.org
bfd7a8c448 Fix compile errors on webrtc/base. 
R=fbarchard@google.com, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 04:59:52 +00:00
andresp@webrtc.org
0229cbae33 Remove ambiguous call to MakeCheckOpString. 
BUG=3777
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7051 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 04:53:29 +00:00