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1599 Commits

Author SHA1 Message Date
jiayl@webrtc.org
d91608dd2d The 2x2 black frame on windows when the shared window is minimized caused an assert from vp8 and may lead to memroy corruption. 
It's changed to 1x1 to match the behavior on Mac. The Chromium code will detect the size and convert it to a black frame in the original size.

TBR=sergeyu@chromium.org
BUG=crbug/414908

Review URL: https://webrtc-codereview.appspot.com/23709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7214 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 16:12:49 +00:00
henrik.lundin@webrtc.org
5422e724d3 Modifying NetEqExternalDecoderTest 
This change is in preparation for extending the test with another
fixture that inherits from NetEqExternalDecoderTest.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7213 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 15:09:08 +00:00
stefan@webrtc.org
5a098c51ea Refactor VP8 de-packetizer. 
It's duplicated to parse VP8 RTP packet at the moment. We firstly call
RTPPayloadParser functions to save parsed information in RTPPayload
structure, then copy them to RTP header.

This CL removes RTPPayloadParser class and directly saves parsed data in
RTP header.

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

Review URL: https://webrtc-codereview.appspot.com/27449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 11:58:20 +00:00
andresp@webrtc.org
3bd5603b18 Revert "Disable video_capture_tests for Android." (revision 7023). 
BUG=3768
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7210 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 11:56:25 +00:00
andresp@webrtc.org
a74eda1b6f Split video_capture_module specific implementation (external vs internal capture) 
into its own targets. Dependencies must link directly with the desired one.

Targets linking with libjingle_media:
 - internal implementation when build_with_chromium=0, default otherwise.

Targets linking with default/external capture implementation:
 - anything dependent on webrtc_test_common
 - anything dependent on video_engine_core

Targets linking with internal capture implementation:
 - vie_auto_test
 - anything dependent on webrtc_test_renderer

GN changes:
 - Not many since there is almost no test definitions.

TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.

BUG=3768
R=glaznev@webrtc.org
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7209 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 11:50:19 +00:00
andresp@webrtc.org
85ef770d92 Split video engine android initialization into each internal module initialization. 
This is to later on allow targets to pick at link time if to include the external or internal implementation. In order to do that the video_engine cannot compile different based on which option is picked later on.

BUG=3768,3770
R=glaznev@webrtc.org, stefan@webrtc.org
TBR=henrike@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7208 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 11:44:51 +00:00
pbos@webrtc.org
ab990ae43a Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."" 
Re-lands r7114 after landing r7204 to adress the compile error causing
the rollback in r7151.

BUG=3070
TBR=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7207 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 09:02:25 +00:00
henrikg@webrtc.org
dae612ebf8 Mark all virtual overrides in the hierarchies of UdpTransportData and 
UdpSocketWrapper 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 removes an unused function.

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

Review URL: https://webrtc-codereview.appspot.com/23579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7195 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 15:29:02 +00:00
bjornv@webrtc.org
c75f607042 audio_processing/aec: Ported NEON optimizations of SubbandCoherence() and its sub-functions to SSE2 
These optimizations were originally committed in r6860, but reverted in r6861, since it broke a bitexactness test (ApmTest.Process) in modules_unittests. That test has now been updated in r7149, hence this CL now pass the test.

BUG=3767
TESTED=manually on linux and trybots
TBR=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7189 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 05:01:42 +00:00
andrew@webrtc.org
6ae5a6d7fe Add a target for the approved subset of rtc_base. 
rtc_base drags in a bunch of unwieldly dependencies (e.g. nss and
json) not required for standalone webrtc (aka rtc/media). The root of
the problem appears to be that MessageQueue depends on a socket server.
(And since common.h -> logging.h -> thread.h -> messagequeue.h, this
dependency spreads quickly.)

This starts a new target for a "purified" subset of rtc_base. It adds
the files which are already being used, replacing the use of common.h
with checks.h. desktop_capture is a lost cause, and retains its
dependency on the full rtc_base.

The hope is that as additional components are desired they will be
cleaned and added to rtc_base_approved.

BUG=3806
R=andresp@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 01:03:29 +00:00
sergeyu@chromium.org
b3cbeb31cc Fix memory leak in webrtc::MouseCursorMonitorMac 
BUG=3815
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/24579004

Patch from Vicken Simonian <vsimon@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7187 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-15 20:11:23 +00:00
bjornv@webrtc.org
7bb2586c55 audio_processing: Correct sample rate in aec_debug_dump 
When writing to wav files in the low level flag aec_debug_dump incorrect sample rates were used for recordings using rates from 32 kHz and above. This since internally inside the AEC we process the data using 16 kHz. Any upper band is processed and combined later on.

This CL adds the correct sample rate to the recording.

BUG=3359
TESTED=locally on 44.1 kHz recordings on Linux
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7182 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-15 13:23:07 +00:00
andresp@webrtc.org
76ba7caae8 Re-enable neteq_performance_unittest.cc for android. 
BUG=3770
R=kjellander@webrtc.org
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7181 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-15 12:29:50 +00:00
glaznev@webrtc.org
91ee7468dd Add enable flag for Android device orientation change event. 
There are reports (not reproducible with appRtcDemo) that
outstanding device orientation change event
OrientationEventListener.onOrientationChanged can be
triggered even after these events are disabled by
OrientationEventListener.disable() code.
Avoid calling native code in this case since underlying
C++ class may have already been deleted.

BUG=3564
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7172 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-12 16:48:12 +00:00
pbos@webrtc.org
1fb5d1204b Initialize restored_packet in nack_rtx_unittest.cc. 
This is to get the DrMemory Full bots to go green, this was previously
suppressed. This fix is likely hiding a real bug that should be
investigated, but it's not a regression from before. The issue should
not be closed before we figure out why this is the case and revert this
"fix".

TBR=stefan@webrtc.org
BUG=3183

Review URL: https://webrtc-codereview.appspot.com/30369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7169 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-12 16:16:00 +00:00
andresp@webrtc.org
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
pbos@webrtc.org
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
pbos@webrtc.org
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
stefan@webrtc.org
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
henrike@webrtc.org
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
jiayl@webrtc.org
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
pbos@webrtc.org
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
aluebs@webrtc.org
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
henrikg@webrtc.org
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
bjornv@webrtc.org
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
henrik.lundin@webrtc.org
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
henrike@webrtc.org
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
glaznev@webrtc.org
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
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
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
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
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
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
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
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
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
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
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