andresp@webrtc.org
e6013bb7be
Fix loopback test for case where no constraint is given.
...
R=stefan@webrtc.org
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5881 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-10 14:21:45 +00:00
asapersson@webrtc.org
2a770828d8
Remove usage of webrtc trace in video processing modules.
...
BUG=3153
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11089005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5880 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-10 11:30:49 +00:00
andresp@webrtc.org
0273fa98e0
Add ability to control peer connection constraints for the loopback test.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11419005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5879 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-10 09:40:16 +00:00
buildbot@webrtc.org
15192f909e
(Auto)update libjingle 64594651-> 64630087
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5878 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-10 06:34:32 +00:00
fischman@webrtc.org
f93021430d
Remove self-assignment hacks that were added to avoid unused variable warnings.
...
Instead, appear to use the variables.
BUG=3152
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5877 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 21:19:55 +00:00
andrew@webrtc.org
0569d93db7
Move a chatty creation log in neteq to LS_VERBOSE.
...
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5876 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 17:48:48 +00:00
henrike@webrtc.org
8f89497949
Remove erronuous commit message.
...
BUG=N/A
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5875 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 14:43:43 +00:00
solenberg@webrtc.org
f4357f3530
Make Android-APK compile in release again.
...
BUG=3152
R=kjellander@webrtc.org
TBR=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5874 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 14:21:37 +00:00
kjellander@webrtc.org
52fd65b16a
Partial revert of "Removing samples directory following move to Github"
...
Reason: Unfortunately we depend on AppRTC being in this location
for the bots in our Chromium WebRTC waterfalls so I'm reverting
this until we've solved that dependency.
This reverts apprtc and adapter.js from being removed in r5871.
R=phoglund@webrtc.org
TBR=dutton@google.com
BUG=
Review URL: https://webrtc-codereview.appspot.com/11529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5873 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 13:52:24 +00:00
henrika@webrtc.org
8883a0f47f
(landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds
...
Landing https://webrtc-codereview.appspot.com/11419004/ manually.
TBR=niklase
BUG=none
Review URL: https://webrtc-codereview.appspot.com/11439005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5872 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 13:04:12 +00:00
dutton@google.com
7ecc142d6b
Removing samples directory following move to Github
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5871 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 09:55:54 +00:00
buildbot@webrtc.org
61c1b8ea32
(Auto)update libjingle 64585415-> 64594651
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5870 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 06:06:38 +00:00
fischman@webrtc.org
2e9d89cf77
Unbreak android APK buildbots by emptying the video_capture_tests_apk target.
...
Needed until the bots start to specify include_internal_video_capture=1.
TBR=henrike@webrtc.org
BUG=3152
Review URL: https://webrtc-codereview.appspot.com/11479006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5869 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 02:34:50 +00:00
fischman@webrtc.org
b0b135e4c2
VideoCaptureAndroid: support multiple frame-rates per resolution.
...
Also enables running video_capture_tests_apk on the WebRTC/Chromium APK bots,
assuming GYP_DEFINES includes include_tests=1 and
include_internal_video_capture=1.
This required running VideoCaptureAndroid's camera capture on a dedicated thread, matching other platform's video_capture impls.
BUG=2974,3152
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5868 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 01:18:32 +00:00
sergeyu@chromium.org
74f6074ec1
Fix DesktopSize::is_empty() for the case when only width or only height is 0.
...
BUG=crbug.com/358909
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/11479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5867 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 01:04:22 +00:00
andrew@webrtc.org
a78a41f985
Move output_mixer_unittest.cc to utility_unittest.cc.
...
This reflects a move of the tested code in:
https://webrtc-codereview.appspot.com/11019005/
TBR=xians
Review URL: https://webrtc-codereview.appspot.com/11449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5866 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-08 23:09:28 +00:00
fischman@webrtc.org
f4c9444c65
VideoCaptureAndroid: stop referencing ViERenderer
...
To facilitate building video_capture's java code without video_render's java
code this reorganizes the local-preview hack to be driven by MediaEngine.
This is the "first step" in the linked bug.
BUG=3175
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5865 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-08 22:55:07 +00:00
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