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Author SHA1 Message Date
henrike@webrtc.org
f824fde36f (Auto)update libjingle 64326665-> 64585415 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5864 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-08 22:13:01 +00:00
fischman@webrtc.org
984e4fbaaa video_capture(iOS): move stopCapture to background thread 
Also suspend frame delivery on stopCapture() to avoid pause+onVideoError
during hangup.

BUG=3162
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/11389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5863 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-08 21:06:52 +00:00
pbos@webrtc.org
2a03498825 Implement FEC support in VideoReceiveStream. 
Added an FEC end-to-end test. NACK+FEC is probably working but not yet tested
as the test for it must introduce packet delays as the underlying API prefers
NACK over FEC if RTT is low.

BUG=3174
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5862 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-08 11:21:45 +00:00
andresp@webrtc.org
dc80bae2a6 Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. 
Clean some logs and add asserts in the way.

BUG=3153
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5861 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-08 11:06:12 +00:00
henrik.lundin@webrtc.org
b287d968d9 New NetEq test to verify correct timestamp propagation 
BUG=3154
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5860 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-07 21:21:45 +00:00
henrike@webrtc.org
74a7c482b9 Removes unused thread causing compiler warnings. 
BUG=N/A
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5859 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-07 20:49:34 +00:00
wu@webrtc.org
4e393070be Compare the answer's media type against offer to make sure they are match. Otherwise we should return failure. 
BUG=2687
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11079005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5858 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-07 17:04:35 +00:00
henrike@webrtc.org
413d001132 Removed the disabling of include_tests from r2729. 
BUG=N/A
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5856 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-07 15:52:31 +00:00
elham@webrtc.org
9337c839da Updated WebRTC version to 3.52 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5855 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-07 15:49:00 +00:00
stefan@webrtc.org
b08db28958 Clean up traces and logs in RemoteBitrateEstimator. 
BUG=3153
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5854 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-07 12:53:28 +00:00
mflodman@webrtc.org
5574dacd1f Log Fixit for parts of video_engine folder. 
BUG=3153
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5853 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-07 10:56:31 +00:00
kjellander@webrtc.org
e8d1865408 Disable more tests for DrMemory to speed up execution. 
Disable a few more tests on Windows when running under
Dr Memory to get the build time down to a reasonable total.

BUG=None
TEST=None
TBR=phoglund

Review URL: https://webrtc-codereview.appspot.com/11299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5852 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-07 09:00:12 +00:00
andresp@webrtc.org
36947bb635 Fix logging calls in bitrate_controller module. 
BUG=3153
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11069005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5851 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-07 08:45:16 +00:00
kjellander@webrtc.org
9f57404334 Excluding and suppressing Dr Memory test failures. 
With these tests excluded and failures suppressed
we should be able to bring Dr Memory Full into a
green state in
http://build.chromium.org/p/client.webrtc.fyi/waterfall
so we can move the bots into the main waterfall.

BUG=3158, 3159
TEST=Ran successful runs of the tests that never completed
using the reproduction steps in the issues listed above on
a local Windows box. The tests that just failed weren't tried,
since they cannot have been blocking other possibly failing
tests in the same binary.

R=pbos@webrtc.org
TBR=pbos

Review URL: https://webrtc-codereview.appspot.com/11209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5850 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-07 08:01:06 +00:00
pbos@webrtc.org
0fefb1041c Remove WEBRTC_TRACE use in common_video/ 
Replaces a NOTREACHED() macro with inline assert(false).

BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5849 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-07 07:29:18 +00:00
henrike@webrtc.org
09b0c10eed Talk: fixes warning: local variable is initialized but not referenced due to only using the variable in question for asserts. 
BUG=N/A
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5848 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-04 22:33:34 +00:00
fischman@webrtc.org
d1fe6b728e AppRTCDemo(android): fix a couple of SDP-related regressions. 
- r5834 made it so that empty fields are a fatal SDP parsing error, exposing
  opportunities for improvement in the preferISAC; changed split/join to use
  \r\n instead of \n and now omitting the trailing space on the m=audio line
  that triggered the new failure.
- DTLS requires a different role for each endpoint so conflicts with loopback
  calling.  apprtc.py suppresses DTLS for that reason in loopback calls, so the
  android demo app now only enables DTLS by default if it is not suppressed by a
  constraint (matching Chrome).

BUG=3164,3165,2507
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5847 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-04 21:40:46 +00:00
jiayl@webrtc.org
f040bd8fa3 Fix a crash in WindowCapturereMac when capture() fails. 
BUG=http://code.google.com/p/chromium/issues/detail?id=359985
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/11219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5846 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-04 20:26:41 +00:00
henrike@webrtc.org
f5bebd40f3 (Auto)update libjingle 64247466-> 64326665 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5845 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-04 18:39:07 +00:00
michaelbai@google.com
653c325af2 Fix the library path for android 64-bit build 
BUG=359687
R=andrew@webrtc.org, fischman@webrtc.org, torne@chromium.org

Review URL: https://webrtc-codereview.appspot.com/11149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5844 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-04 04:44:19 +00:00
andrew@webrtc.org
40ee3d07ed Consolidate audio conversion from Channel and TransmitMixer. 
Replace the two versions with a single DownConvertToCodecFormat. As
mentioned in comments, this could be further consolidated with
RemixAndResample but we should write a full audio converter class in
that case.

Along the way:
- Fix the bug present in Channel::Demultiplex with mono input and a
stereo codec.
- Remove the 32 kHz max from the OnDataAvailable path. This avoids a
48 -> 32 -> 48 conversion when VoE is passed 48 kHz audio; instead we
get a straight pass-through to ACM. The 32 kHz conversion is still
needed in the RecordedDataIsAvailable path until APM natively supports
48 kHz.
- Merge resampler improvements from ACM1 to ACM2. This allows ACM to
handle 44.1 kHz audio passed to VoE and was originally done here:
https://webrtc-codereview.appspot.com/1590004
- Reuse the RemixAndResample unit tests for DownConvertToCodecFormat.
- Remove unused functions from utility.cc.

BUG=3155,3000,b/12867572
TESTED=voe_cmd_test using both the OnDataAvailable and
RecordedDataIsAvailable paths, with a captured audio format of all
combinations of {44.1,48} kHz and {1,2} channels, running through all
codecs, and finally using both ACM1 and ACM2.

R=henrika@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11019005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5843 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-03 21:56:01 +00:00
dutton@google.com
cca888a5bf Removed rehydrate.html 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5842 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-03 21:25:54 +00:00
andrew@webrtc.org
be8e8ee6f6 Remove bad *s from filename. 
Appeared to be causing an error on the Windows bots:
svn: Can't check path
'E:\b\build\slave\win\build\src\samples\js\demos\html\****THESE_FILES_ARE_MOVING****':
The filename, directory name, or volume label syntax is incorrect.

TBR=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/11069006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5840 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-03 20:51:41 +00:00
kjellander@webrtc.org
c7b8b2f2a7 PRESUBMIT.py: use new way to specify default try builders 
In https://codereview.chromium.org/178223016 and
https://codereview.chromium.org/197963003 the way the
PRESUBMIT.py specifies the default try builders for a
try job have changed.

When submitting a try job now, the test filter argument no
longer works unless --bot is also specified.
This CL attempts to resolve this by moving away from the
deprecated approach onto using the new format instead.

This CL also includes two new trybots: win_asan and linux_tsan2
(added in https://codereview.chromium.org/220453004).

BUG=3148
TEST=Successfully fired off a -t compile job where the
test filter worked.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5839 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-03 20:19:36 +00:00
dutton@google.com
fe165ded46 Added warning for Github move ****THESE_FILES_ARE_MOVING**** 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5837 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-03 19:57:06 +00:00
bjornv@webrtc.org
240eec3cd4 Delay Estimator: Minor refactoring and added a setter function. 
* Replaced the lookahead input parameter at Create() with a setter. This makes it slightly more user friendly.
* Changed the buffer shifting in SoftReset... to become more readable.

TESTED=trybots, modules_unittests
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5836 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-03 08:11:47 +00:00
wu@webrtc.org
148149138d (Auto)update libjingle 64147530-> 64247466 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 23:25:15 +00:00
wu@webrtc.org
5e760e7b94 Check the return value of the FromString call and return failure when then value is invalid. I.e. uses 
bool FromString(const std::string& s, T* t)
instead of
T FromString(const std::string& str)

Before this change we will silently continue the parsing and take whatever default value returned by FromString.

TEST=new tests
BUG=2507
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5834 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 23:19:09 +00:00
wu@webrtc.org
e387771b98 Remove webrtc_unittest.cc from talk presubmit script. 
BUG=
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5833 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 22:23:16 +00:00
henrik.lundin@webrtc.org
184b913eb5 Rename RTPanalyze to rtp_analyze and remove old version 
The tool RTPanalyze (used to process an input RTP dump into a
text file of RTP header info) was present in both the neteq and
neteq4 folders. This change pulls in changes from the old to the new
and renames the source file and tool to rtp_analyze.

Removing special code for dummy-rtp files (it is supported without
special code), and making the RED payload type settable using flags.

Moving from test/ to tools/ folder.

BUG=2692
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5832 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 20:56:17 +00:00
andrew@webrtc.org
c7c432aa9b Remove AudioDevice::{Microphone,Speaker}IsAvailable. 
This was only used for logging, except on Mac, where the methods are
now private.

BUG=3132
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5831 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 16:49:26 +00:00
minyue@webrtc.org
7549ff4257 This is to get rid of a bug relating to the return of NULL in calling GetDecoder when there are DTMF packets. 
BUG=3140
TEST=trybots
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10929006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5830 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 15:03:01 +00:00
henrik.lundin@webrtc.org
1092ea0192 Add format specification to output file names 
This change facilitates running ApmTest.VerifyDebugDumpInt and
ApmTest.VerifyDebugDumpFloat in parallel, since they are not writing
to the same files any longer.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5829 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 07:46:49 +00:00
henrika@webrtc.org
620d444c0b Extends max sample rate from 96kHz to 192kHz on the input side. 
TEST=apprtc in Chrome using this WebRTC version and a device on Windows which can capture at 192kHz
BUG=725
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5828 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 07:22:34 +00:00
braveyao@webrtc.org
790385fee4 sink_filter_ds.cc: add lock to Receive procedure to Pause(). 
BUG=2233
TEST=AUTO Test
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5827 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 02:14:55 +00:00
andrew@webrtc.org
19018ddb17 Make ACM2 the default in voe_cmd_test. 
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5826 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 20:58:05 +00:00
wu@webrtc.org
05e7b44b83 (Auto)update libjingle 63948945-> 64147530 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5825 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 17:44:24 +00:00
stefan@webrtc.org
f8f7c8b618 Added simulations of capacity variations and wifi recordings. 
Also changes the packet sizes for the video sender and the trace based filter to match.

R=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5824 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 14:00:05 +00:00
kjellander@webrtc.org
7e889b7126 Add /third_party/syzygy/binaries to .gitignore 
This should have been done in
https://webrtc-codereview.appspot.com/2381004

BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5823 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 13:46:48 +00:00
kjellander@webrtc.org
d10bdd3f78 Roll chromium_revision 255773:260462 
This disables GN use for the moment (Chromium
has disabled it for now but plan to pick up the
work at a later stage). I'm leaving the rest of
the GN stuff in our DEPS since that's how
the Chromium DEPS currently looks like.

Overview of changes in Chrome DEPS:
$ svn diff http://src.chromium.org/chrome/trunk/src/DEPS -r 255773:260462

which can be compared with the output of:
$ svn cat http://webrtc.googlecode.com/svn/trunk/DEPS | grep chromium_deps | sed 's/^ *//' | sort | uniq

in a WebRTC checkout, gives the following relevant changes:
* third_party/android_tools 0582bd:ca3567
* third_party/icu 249466:259309
* third_party/libjpeg_turbo 251747:259851
* third_party/libyuv 979:986
* third_party/nss 254867:259440
* tools/gyp 1860:1880

The following also shows that Clang is upgraded from r198389 to r202554:
$ svn diff http://src.chromium.org/chrome/trunk/src/tools/clang/scripts/update.sh -r 255773:260462

TEST=trybots
BUG=None
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5822 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 10:40:03 +00:00
andrew@webrtc.org
ca9d038ac8 Fix ARM64 detection. 
Use only __aarch64__ and don't look for __arm64__ at all.
It turns out that clang defines both and GCC only the former.
Hence, looking only for __aarch64__ should be safe.

BUG=chromium:354405,chromium:358092
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10939004

Patch from Primiano Tucci <primiano@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5821 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 01:19:08 +00:00
fischman@webrtc.org
a789f3720a VoiceEngine(iOS & Android): removed NOT_SUPPORTED 
Also:
- removed underflow of a uint32 creating crazy-large delay values
- removed always-fail AudioDeviceIPhone::MicrophoneIsAvailable() impl (see
  bug 3132)
- removed unnecessary exclusion of features from iOS & Android builds

BUG=2050,3132
R=andrew@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10909005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5820 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 00:16:35 +00:00
fbarchard@google.com
8f8119409d Roll libyuv to 994 for arm64 initial support using C versions of code. 
BUG=chromium:354539
TESTED=GYP_DEFINES="OS=ios target_arch=armv7 target_subarch=64" GYP_CROSSCOMPILE=1 GYP_GENERATOR_FLAGS="output_dir=out_ios" ./build/gyp_chromium -f ninja --depth=. libyuv_test.gyp  && ninja -j7 -C out_ios/Debug-iphoneos
R=andrew@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/10929005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5819 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-31 21:35:01 +00:00
fischman@webrtc.org
49c5ba32bb AppRTCDemo(iOS): now works in the iOS Simulator! 
...which has no camera device emulation or pass-through, so no local video
view.

R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5815 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-31 20:22:19 +00:00
fischman@webrtc.org
61e78fca6c AppRTCDemo(iOS): remote-video reliability fixes 
Previously GAE Channel callbacks would be handled by JS string-encoding the
payload into a URL.  Unfortunately this is limited to the (undocumented,
silently problematic) maximum URL length UIWebView supports.  Replaced this
scheme by a notification from JS to ObjC and a getter from ObjC to JS (which
happens out-of-line to avoid worrying about UIWebView's re-entrancy, or lack
thereof).  Part of this change also moved from a combination of: JSON,
URL-escaping, and ad-hoc :-separated values to simply JSON.

Also incidentally:
- Removed outdated TODO about onRenegotiationNeeded, which is unneeded
- Move handling of PeerConnection callbacks to the main queue to avoid having
  to think about concurrency too hard.
- Replaced a bunch of NSOrderedSame with isEqualToString for clearer code and
  not having to worry about the fact that [nil compare:@"foo"]==NSOrderedSame
  is always true (yay ObjC!).
- Auto-scroll messages view.

BUG=3117
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10899006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-31 20:16:49 +00:00
fbarchard@google.com
30cd5b5278 libyuv roll to r986 for c89 fix to cpu_id. 
BUG=none
TESTED=cl cpu_id.cc
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-31 17:28:46 +00:00
solenberg@webrtc.org
caeae4680c Add tests for the RBE RemoveStream() API. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5812 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-31 13:33:39 +00:00
henrik.lundin@webrtc.org
d0a81d91ff VoE Channel: Don't register codecs when stopping receiver 
VoiceEngine's Channel::StopReceiving() would call
RegisterReceiveCodecsToRTPModule(), which caused some errors
with RED and ULP-FEC. In particular, an error message would be
printed when hanging up a call in voe_cmd_test application.

BUG=3085
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5811 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-31 07:31:53 +00:00
fischman@webrtc.org
fe16488184 AppRTCDemo(android): specify DtlsSrtpKeyAgreement:true in CreatePeerConnection's constraints. 
This is required to interop with Chrome now that SDES is disabled in Chrome (as of r5640).

BUG=2774
R=jiayl@chromium.org

Review URL: https://webrtc-codereview.appspot.com/10749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5809 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-28 19:58:03 +00:00
fischman@webrtc.org
4f2bd68744 Silence pointless LS_WARNING about port 0 for active-only candidates. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5808 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-28 18:13:34 +00:00