d2cf48de1a
Fix mac video_render implementation on cocoa.
...
Hit this while playing around with all compile possibilities for issue 3770.
BUG=3770
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7166 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-12 13:57:47 +00:00
f7e5f22f98
Fix stack limit exceeded in http client.
...
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7165 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-12 13:35:05 +00:00
a0d7827b16
Add ability to downscale content to improve quality.
...
BUG=3712
R=marpan@google.com , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7164 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-12 11:51:47 +00:00
b5e6bfc76a
Make RTPSender/RTPReceiver generic.
...
Changes include,
1) Introduce class RtpPacketizerGeneric & RtpDePacketizerGeneric.
2) Introduce class RtpDepacketizerVp8.
3) Make RTPSenderVideo::SendH264 generic and used by all packetizers.
4) Move codec specific functions from RTPSenderVideo/RTPReceiverVideo to
RtpPacketizer/RtpDePacketizer sub-classes.
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26399004
Patch from Changbin Shao <changbin.shao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7163 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-12 11:05:55 +00:00
6071b0636d
Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such.
...
This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.
This also highlighted a number of unused functions which I've removed.
-- This is was reviewed in https://webrtc-codereview.appspot.com/19309004/ , but
-- a new cl was needed to resolve a small conflict before committing.
BUG=none
TEST=none
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7162 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-12 07:42:33 +00:00
cc774a69cb
Mark all virtual overrides in the hierarchies of RtpDump and
...
VCMPacketizationCallback as such.
This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.
This also marks all other such overrides in the affected files.
BUG=none
TEST=none
R=henrike@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7161 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 22:45:54 +00:00
89959966a9
Fix window capturing on Windows when the window is minimized.
...
BUG=crbug/410290
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/20319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7158 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 19:33:58 +00:00
f520ea5eed
Skip dlclose() on AddressSanitizer.
...
AddressSanitizer can't symbolize parts of the stack that contains
dlclose()d modules. This makes some LSan suppressions not kick in and
blocks launching the LSan bot for WebRTC.
This "fix" excludes dlclose() in
webrtc/modules/audio_device/linux/latebindingsymboltable_linux.cc which
resolves this on the bot.
R=xians@webrtc.org
BUG=3402,chromium:375154
Review URL: https://webrtc-codereview.appspot.com/25499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7157 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 17:29:11 +00:00
b9906743da
Split suppressons of thread.cc and messagequeue.cc.
...
Most calls have either of these in the stack, meaning that pretty much
all races are suppressed.
BUG=
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7154 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 14:59:06 +00:00
4b049fcabe
Remove developing code in ns_core
...
This defines were hardcoded and the code inside of the ifdefs was never used.
BUG=webrtc:3763
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7153 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 11:19:56 +00:00
307d3dbdee
Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."
...
Speculative revert, seems to be reason for flaky Win FYI bot compile break.
> 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
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7151 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 09:48:30 +00:00
665d861115
Restore webrtc_base target until r7140 is rolled into Chromium.
...
In r7140 the webrtc_base target was renamed to rtc_base. This
breaks our FYI bots for rolling WebRTC in Chromium's DEPS.
By re-adding a None target named webrtc_base, this transition
should be smoother.
TBR=henrikg@webrtc.org ,
TESTED=Passed build/gyp_chromium on a Chromium checkout with src/third_party/webrtc replaced by a mount like this:
cd /path/to/chromium/src
sudo mount --bind /path/to/webrtc/trunk/webrtc third_party/webrtc
Review URL: https://webrtc-codereview.appspot.com/23589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7150 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 09:22:13 +00:00
8dd60cc855
audio_processing_unittests: Enabled ApmTest.Process for all platforms but Android
...
During porting some neon optimizations to sse2 the ApmTest.Process failed despite bit exact outputs. The reason is that with float data between component, as to previously truncating to int, we get small deviations in logged metrics. This affected a noise probability to a small fraction, which is not a particular bug.
This CL change the comparison from EXPECT_EQ() to EXPECT_NEAR() which then as a result makes the test run on Mac and Windows as well.
For int values a deviation of 1 is acceptable, which would include any rounding errors.
For float values a deviation of 0.0005 is chosen by looking at current test stats for the affected platforms/optimizations.
BUG=114
TESTED=locally on linux with and without sse2 optimizations and trybots
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7149 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 08:36:35 +00:00
2b58a4433f
Calculating round-trip-time in send-only channel in VoE.
...
TESTS=built chromium and tested with 1:1 hangout call
BUG=
R=stefan@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7147 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 07:51:53 +00:00
1972ff8a6e
Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.
...
This will make a subsequent change I intend to do safer, where I'll change the
return type of one of the base Module functions, by breaking the compile if I
miss any overrides.
This also highlighted a number of unused functions (in many cases apparently
virtual "overrides" of no-longer-existent base functions). I've removed some of
these.
This also highlighted several cases where "virtual" was used unnecessarily to
mark a function that was only defined in one class. Removed "virtual" in those
cases.
BUG=none
TEST=none
R=andrew@webrtc.org , henrik.lundin@webrtc.org , mallinath@webrtc.org , mflodman@webrtc.org , stefan@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7146 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 06:20:28 +00:00
47658f1269
Mark all virtual overrides in the hierarchy of AudioPacketizationCallback,
...
RTPStream, and NetEq as such. Also mark all other virtual overrides in the same
files.
This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.
This also deletes ACMTest.cc, which existed solely to define ~ACMTest(), which
was marked pure virtual in the header. (Pure virtual destructors still need a
definition.) Because there is another pure virtual method in this class, the
class is already abstract, so there's no benefit to making the desturctor pure.
Making it non-pure allows removing the separate source file.
BUG=none
TEST=none
R=henrik.lundin@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7144 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 22:14:59 +00:00
1711104b8a
Fix MSVC warnings about value truncations, webrtc/base/ edition.
...
BUG=chromium:81439
TEST=none
R=henrike@webrtc.org , marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/20249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7143 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 22:10:24 +00:00
3472dcd7b0
Fix frame rate selection for Android camera.
...
- Android camera supports multiple fps values for a single video
resolution - change video source default video format selection
to pick up best available fps.
- Change fps range calculation to better match target fps value.
BUG=2622
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7142 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 19:24:57 +00:00
67eabc0938
Add schannel webrtc_base build using a new use_schannel gyp variable.
...
R=henrike@webrtc.org , thorcarpenter@google.com
Review URL: https://webrtc-codereview.appspot.com/28409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7141 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 18:06:47 +00:00
b2efb6771c
Put base tests in webrtc_tests.gyp
...
BUG=N/A
R=andrew@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7140 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 17:28:19 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
a781f68712
Fix rm command for class cleanup in r7091
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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
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
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
