3672 Commits

Author SHA1 Message Date
henrik.lundin@webrtc.org
e9217b4bdb Remove WebRtcACMEncodingType
The parameter was not needed; it was sufficient with a bool indicating
speech or not speech. This change propagates to the InFrameType
callback function. Some tests are updated too.

COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org
TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42209004

Cr-Commit-Position: refs/heads/master@{#8626}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8626 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 07:51:21 +00:00
marpan@webrtc.org
16a87b97f9 Add VP9 denoiser test to videoprocessor_integrationtest.
TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/43599004

Cr-Commit-Position: refs/heads/master@{#8622}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8622 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 22:19:15 +00:00
aluebs@webrtc.org
1d88394bcb Add support for arbitrary array geometries in Beamformer
R=andrew@webrtc.org, mgraczyk@chromium.org

Review URL: https://webrtc-codereview.appspot.com/38299004

Cr-Commit-Position: refs/heads/master@{#8621}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8621 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 20:39:20 +00:00
bjornv@webrtc.org
d7a212e8b9 audio_processing/aec: Increased delay metrics aggregation window to five seconds
The known clients (GetStats and UMA histogram in Chrome) use at least 5 second aggregation window. There is no particular value in calculating the metrics more often.

The CL also includes a small refactoring moving a declaration inside an if statement.

BUG=2994
TEST=N/A
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40219004

Cr-Commit-Position: refs/heads/master@{#8619}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8619 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 16:14:58 +00:00
stefan@webrtc.org
c3f15c08bc Fix scoped_ptrs in bwe_simulations.
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45469004

Cr-Commit-Position: refs/heads/master@{#8618}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8618 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 16:06:21 +00:00
magjed@webrtc.org
2386d6dd92 Revert 8599 "Revert 8580 "Unify underlying frame buffer in I420VideoFrame and...""
It's possible to build Chrome on Windows with this patch now.

BUG=1128

> This is unfortunately causing build problems in Chrome on Windows.

>> Unify underlying frame buffer in I420VideoFrame and WebRtcVideoFrame
>>
>> Currently, I420VideoFrame uses three webrtc::Plane to store pixel data, and WebRtcVideoFrame uses WebRtcVideoFrame::FrameBuffer/webrtc::VideoFrame. The two subclasses WebRtcTextureVideoFrame and TextureVideoFrame use a NativeHandle to store pixel data, and there is also a class WebRtcVideoRenderFrame that wraps an I420VideoFrame.
>>
>> This CL replaces these classes with a new interface VideoFrameBuffer that provides the common functionality. This makes it possible to remove deep frame copies between cricket::VideoFrame and I420VideoFrame.
>>
>> Some additional minor changes are:
>> * Disallow creation of 0x0 texture frames.
>> * Remove the half-implemented ref count functions in I420VideoFrame.
>> * Remove the Alias functionality in WebRtcVideoFrame
>>
>> The final goal is to eliminate all frame copies, but to limit the scope of this CL, some planned changes are postponed to follow-up CL:s (see planned changes in https://webrtc-codereview.appspot.com/38879004, or https://docs.google.com/document/d/1bxoJZNmlo-Z9GnQwIaWpEG6hDlL_W-bzka8Zb_K2NbA/preview). Specifically, this CL:
>> * Keeps empty subclasses WebRtcTextureVideoFrame and TextureVideoFrame, and just delegates the construction to the superclass.
>> * Keeps the deep copies from cricket::VideoFrame to I420VideoFrame.
>>
>> BUG=1128
>> R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org, tommi@webrtc.org
>>
>> Review URL: https://webrtc-codereview.appspot.com/42469004

R=pbos@webrtc.org
TBR=mflodman, pbos, perkj, tommi

Review URL: https://webrtc-codereview.appspot.com/45489004

Cr-Commit-Position: refs/heads/master@{#8616}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8616 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 14:03:51 +00:00
pbos@webrtc.org
67a9e40286 Prevent encoding frames with wrong resolution.
This is a speculative fix for a crash that should be able to happen if a
codec is reconfigured while a frame is leaving the
VideoProcessingModule, causing a mismatch between configured codec and
input frame size.

BUG=
R=magjed@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48379004

Cr-Commit-Position: refs/heads/master@{#8615}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8615 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 13:58:16 +00:00
tommi@webrtc.org
03054486f5 Adding basic support for posting tasks to a process thread.
BUG=
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41099004

Cr-Commit-Position: refs/heads/master@{#8614}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8614 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 13:14:19 +00:00
tommi@webrtc.org
658d2015f3 Allow VideoSender to be constructed on one thread but initialized and used for doing registrations, on another.
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42219004

Cr-Commit-Position: refs/heads/master@{#8613}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8613 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 12:22:22 +00:00
andrew@webrtc.org
fa67463d37 skip isac_neon if neon is not supported
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39909004

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

Cr-Commit-Position: refs/heads/master@{#8610}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8610 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 06:07:51 +00:00
guoweis@webrtc.org
4536289353 Add CVO support to RTP sender side.
According to http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ts_126114v120700p.pdf,
CVO byte should only be added in the last packet of each key frame or when the rotation changes. Currently, we're adding this byte in each frame to start with.

BUG=4145
R=mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42439004

Cr-Commit-Position: refs/heads/master@{#8606}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8606 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 22:55:43 +00:00
marpan@webrtc.org
6daacbc8ae Set cpu_speed parameter for low resolutions, for non-simulcast.
Allow for setting different cpu_speed setting based on resolution, for non-simulcast.
Use the existing low resolution simulcast cpu_speed setting for the non-simulcast case.

No change to simulcast behavior, unless top/highest layer stream is also below CIF resolution,
(in which case all layers will use lower the cpu_speed setting =-4).

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37319004

Cr-Commit-Position: refs/heads/master@{#8603}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8603 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 21:47:18 +00:00
tommi@webrtc.org
1f94407319 Revert 8580 "Unify underlying frame buffer in I420VideoFrame and..."
This is unfortunately causing build problems in Chrome on Windows.

> Unify underlying frame buffer in I420VideoFrame and WebRtcVideoFrame
> 
> Currently, I420VideoFrame uses three webrtc::Plane to store pixel data, and WebRtcVideoFrame uses WebRtcVideoFrame::FrameBuffer/webrtc::VideoFrame. The two subclasses WebRtcTextureVideoFrame and TextureVideoFrame use a NativeHandle to store pixel data, and there is also a class WebRtcVideoRenderFrame that wraps an I420VideoFrame.
> 
> This CL replaces these classes with a new interface VideoFrameBuffer that provides the common functionality. This makes it possible to remove deep frame copies between cricket::VideoFrame and I420VideoFrame.
> 
> Some additional minor changes are:
> * Disallow creation of 0x0 texture frames.
> * Remove the half-implemented ref count functions in I420VideoFrame.
> * Remove the Alias functionality in WebRtcVideoFrame
> 
> The final goal is to eliminate all frame copies, but to limit the scope of this CL, some planned changes are postponed to follow-up CL:s (see planned changes in https://webrtc-codereview.appspot.com/38879004, or https://docs.google.com/document/d/1bxoJZNmlo-Z9GnQwIaWpEG6hDlL_W-bzka8Zb_K2NbA/preview). Specifically, this CL:
> * Keeps empty subclasses WebRtcTextureVideoFrame and TextureVideoFrame, and just delegates the construction to the superclass.
> * Keeps the deep copies from cricket::VideoFrame to I420VideoFrame.
> 
> BUG=1128
> R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org, tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/42469004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42199005

Cr-Commit-Position: refs/heads/master@{#8599}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8599 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 17:35:00 +00:00
henrik.lundin@webrtc.org
c86bbbaa93 Add speech flag to EncodedInfo
The flag indicates if the encoded bitstream is speech or comfort noise.

COAUTHOR=kwiberg@webrtc.org
R=jmarusic@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42629004

Cr-Commit-Position: refs/heads/master@{#8598}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8598 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 16:03:19 +00:00
kjellander@webrtc.org
14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 13:04:54 +00:00
stefan@webrtc.org
792f1a14e2 Break out allocation from BitrateController into a BitrateAllocator.
This also refactors some of the padding and allocation code in ViEEncoder, and
makes ChannelGroup a simple forwarder from BitrateController to
BitrateAllocator.

This CL is part of a bigger picture, see https://review.webrtc.org/35319004/ for
details.

BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org, sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44399004

Cr-Commit-Position: refs/heads/master@{#8595}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8595 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 12:25:17 +00:00
henrik.lundin@webrtc.org
61c22aca5f Eliminate AcmGenericCodec::Add10MsData
All encoding work is now done in the Encode function.

Note: This CL leaves a technical debt in
AudioCodingModuleImpl::Add10MsData. This will be fixed in later
changes.

COAUTHOR=kwiberg@webrtc.org
R=jmarusic@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46419004

Cr-Commit-Position: refs/heads/master@{#8594}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8594 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 11:52:17 +00:00
kjellander@webrtc.org
6dab6d700d Let Chromium declare the mips_dsp_rev build variable.
In https://codereview.chromium.org/883253003, the mips_dsp_rev
build variable is added to Chromium's GYP and GN build files. Remove
the declarations of mips_dsp_rev from WebRTC's GYP and GN build files.

Replace mips_fpu with mips_float_abi and remove the compiler flags that
are already set by Chromium.

The main review of this was done in https://webrtc-codereview.appspot.com/39779004
but since that CL wasn't created with the right base URL, I made
this in order to be able to run WebRTC trybots properly.

BUG=446234
TBR=wtc@chromium.org

Review URL: https://webrtc-codereview.appspot.com/44549004

Cr-Commit-Position: refs/heads/master@{#8590}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8590 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 09:51:17 +00:00
henrik.lundin@webrtc.org
1d25c87199 Reland r8577 "Collapse AudioEncoderDecoderIsacRed into ..."
This effectively reverts r8578.

TBR=jmarusic@webrtc.org

Original commit message:
Collapse AudioEncoderDecoderIsacRed into AudioEncoderDecoderIsac

With this change, support for iSAC-RED is incorporated into the
regular AudioEncoderDecoderIsac class.

COAUTHOR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44539004

Cr-Commit-Position: refs/heads/master@{#8589}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8589 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 08:55:42 +00:00
magjed@webrtc.org
c8895aa2f3 Unify underlying frame buffer in I420VideoFrame and WebRtcVideoFrame
Currently, I420VideoFrame uses three webrtc::Plane to store pixel data, and WebRtcVideoFrame uses WebRtcVideoFrame::FrameBuffer/webrtc::VideoFrame. The two subclasses WebRtcTextureVideoFrame and TextureVideoFrame use a NativeHandle to store pixel data, and there is also a class WebRtcVideoRenderFrame that wraps an I420VideoFrame.

This CL replaces these classes with a new interface VideoFrameBuffer that provides the common functionality. This makes it possible to remove deep frame copies between cricket::VideoFrame and I420VideoFrame.

Some additional minor changes are:
* Disallow creation of 0x0 texture frames.
* Remove the half-implemented ref count functions in I420VideoFrame.
* Remove the Alias functionality in WebRtcVideoFrame

The final goal is to eliminate all frame copies, but to limit the scope of this CL, some planned changes are postponed to follow-up CL:s (see planned changes in https://webrtc-codereview.appspot.com/38879004, or https://docs.google.com/document/d/1bxoJZNmlo-Z9GnQwIaWpEG6hDlL_W-bzka8Zb_K2NbA/preview). Specifically, this CL:
* Keeps empty subclasses WebRtcTextureVideoFrame and TextureVideoFrame, and just delegates the construction to the superclass.
* Keeps the deep copies from cricket::VideoFrame to I420VideoFrame.

BUG=1128
R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42469004

Cr-Commit-Position: refs/heads/master@{#8580}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8580 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 21:22:26 +00:00
henrik.lundin@webrtc.org
bcef431902 Revert r8577 "Collapse AudioEncoderDecoderIsacRed into ..."
Some of the build bots seems to have reacted to this change.

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42169004

Cr-Commit-Position: refs/heads/master@{#8578}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8578 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 20:13:48 +00:00
henrik.lundin@webrtc.org
1fc28f2305 Collapse AudioEncoderDecoderIsacRed into AudioEncoderDecoderIsac
With this change, support for iSAC-RED is incorporated into the regular
AudioEncoderDecoderIsac class.

COAUTHOR=kwiberg@webrtc.org
R=jmarusic@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43549004

Cr-Commit-Position: refs/heads/master@{#8577}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8577 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 19:31:17 +00:00
sprang@webrtc.org
b144b4b74e Fixed bug in SendTimeHistory, where deleting packets via the getter
would not update the oldest suence number.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42589004

Cr-Commit-Position: refs/heads/master@{#8574}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8574 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 15:44:54 +00:00
minyue@webrtc.org
0561716ae2 Adding Opus DTX support in ACM.
This solution does not use the existing VAD/DTX logic of ACM, since Opus DTX is codec feature, while ACM VAD/DTX is mainly for setting the WebRTC VAD/DTX.

During the development of this CL, two old bugs were found and are fixed in this CL too.

They are in
webrtc/modules/audio_coding/test/Channels.cc
and webrtc/modules/audio_coding/main/acm2/acm_opus_unittest.cc
respectively.

BUG=webrtc:1014
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38469004

Cr-Commit-Position: refs/heads/master@{#8573}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8573 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 12:03:14 +00:00
minyue@webrtc.org
db93b68031 Removing NetEq's direct dependencies on Opus headers.
Neteq had a direct dependency on Chromium/third_party/opus. This should be relayed by target webrtc_opus.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42529004

Cr-Commit-Position: refs/heads/master@{#8567}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8567 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 09:28:53 +00:00
aluebs@webrtc.org
c9ce07ed87 Add Config option to enable 48kHz support in AudioProcessing
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45389004

Cr-Commit-Position: refs/heads/master@{#8563}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8563 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 20:07:51 +00:00
magjed@webrtc.org
97ed2a4b70 I420VideoFrame: Remove function ResetSize
This is a partial reland of https://webrtc-codereview.appspot.com/39939004/.

The original CL was reverted because ViECapturer use ResetSize/IsZeroSize on |captured_frame_| as a check to make sure each captured frame is only delivered once. Removing ResetSize introduced a race condition where a captured frame could be delivered multiple times.

I have fixed this problem in this CL by replacing ResetSize with scoped_ptr::release.

BUG=4352
R=perkj@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39359004

Cr-Commit-Position: refs/heads/master@{#8561}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8561 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 17:33:41 +00:00
bjornv@webrtc.org
976c0f3043 audio_processing/aec: NEON code should not be invoked if it is detectable, but is not NEON
There exist devices with runtime checks for NEON, but where the device is not NEON. One such device is Tegra2 on which currently NEON code is running.

This fix adds a missing feature check when initializing the AEC.

BUG=4304
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42159004

Cr-Commit-Position: refs/heads/master@{#8559}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8559 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 16:25:51 +00:00
marpan@webrtc.org
3fe17d1598 Adjust a few thresholds for VP9 tests.
Needed for the upcoming libvpx roll.

TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/44479004

Cr-Commit-Position: refs/heads/master@{#8557}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8557 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 15:34:19 +00:00
magjed@webrtc.org
fd33293d58 I420VideoFrame: Remove functions set_width and set_height
This is a partial reland of https://webrtc-codereview.appspot.com/39939004/.

The functions set_width and set_height in I420VideoFrame are not needed and just add complexity.

R=perkj@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41009004

Cr-Commit-Position: refs/heads/master@{#8556}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8556 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 13:57:44 +00:00
andresp@webrtc.org
e8f50df6b9 Remove avi recorder and corresponding enable_video flags.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42099004

Cr-Commit-Position: refs/heads/master@{#8554}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8554 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 13:07:44 +00:00
henrik.lundin@webrtc.org
f56c162310 Remove AudioCodingModule::Process()
An earlier change moved the encoding work from Process to
Add10MsData; process was just a no-op.

BUG=3520
COAUTHOR=kwiberg@webrtc.org
R=henrika@webrtc.org, minyue@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43439004

Cr-Commit-Position: refs/heads/master@{#8553}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8553 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 12:30:19 +00:00
sprang@webrtc.org
f35e4bc694 Introduce a send time history class, keeping track of packet send times.
BUG=4308
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39229004

Cr-Commit-Position: refs/heads/master@{#8546}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8546 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 09:06:17 +00:00
bjornv@webrtc.org
2f6ae0de5b audio_coding/codec/ilbc: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT
The macro is defined as
#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
    (WEBRTC_SPL_MUL_16_16(a, b) >> (c))

where the latter macro is in C defined as
(For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,mips}.h)

The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
- minor cleanups like remove of unnecessary parantheses and style changes

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

Review URL: https://webrtc-codereview.appspot.com/39139004

Cr-Commit-Position: refs/heads/master@{#8544}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8544 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-01 19:51:31 +00:00
magjed@webrtc.org
7400e0b876 Revert "I420VideoFrame: Remove functions set_width, set_height, and ResetSize"
This reverts commit r8434.

Reason for revert: Introduced a race condition. If ViECaptureProcess() -> SwapCapturedAndDeliverFrameIfAvailable() is called twice without a call to OnIncomingCapturedFrame() in between (with both captured_frame_ and deliver_frame_ populated), an old frame will be delivered again, since captured_frame_->IsZeroSize() will never be true.

BUG=4352
TBR=perkj@webrtc.org, stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40129004

Cr-Commit-Position: refs/heads/master@{#8530}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8530 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-27 15:19:18 +00:00
tommi@webrtc.org
3985f0151a ProcessThread improvements.
* Added a way to notify a Module that it's been attached to a ProcessThread.
  The benefit of this is to give the module a way to wake up the thread
  when it needs work to happen on the worker thread, immediately.
  Today, module instances are typically registered with a process thread
  outside the control of the modules themselves.  I.e. they typically
  don't know about the process thread they're attached to.

* Improve ProcessThread's WakeUp algorithm to not call TimeUntilNextProcess
  when a WakeUp call is requested.  This is an optimization for the above
  case which avoids the module having to acquire a lock or do an interlocked
  operation before calling WakeUp(), which would ensure the module's
  TimeUntilNextProcess() implementation would return 0.

BUG=2822
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39239004

Cr-Commit-Position: refs/heads/master@{#8527}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8527 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-27 13:37:25 +00:00
jmarusic@webrtc.org
abbdd520b0 AudioEncoder: documentation fix
Follow-up to https://webrtc-codereview.appspot.com/38279004/

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38309004

Cr-Commit-Position: refs/heads/master@{#8524}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8524 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-27 09:20:25 +00:00
aluebs@webrtc.org
3aca0b0b31 Add 48kHz support to Beamformer
Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz
Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.

BUG=webrtc:3146
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35159004

Cr-Commit-Position: refs/heads/master@{#8522}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8522 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 21:53:00 +00:00
jmarusic@webrtc.org
b1f0de30be AudioEncoder: change Encode and EncodeInternal return type to void
After code cleanup done on issues:
https://webrtc-codereview.appspot.com/34259004/
https://webrtc-codereview.appspot.com/43409004/
https://webrtc-codereview.appspot.com/34309004/
https://webrtc-codereview.appspot.com/34309004/
https://webrtc-codereview.appspot.com/36209004/
https://webrtc-codereview.appspot.com/40899004/
https://webrtc-codereview.appspot.com/39279004/
https://webrtc-codereview.appspot.com/42099005/
and the similar work done for AudioEncoderDecoderIsacT,  methods AudioEncoder::Encode and AudioEncoder::EncodeInternal will always succeed. Therefore, there is no need for them to return bool value that represents success or failure.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38279004

Cr-Commit-Position: refs/heads/master@{#8518}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8518 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 15:38:46 +00:00
kwiberg@webrtc.org
00b8f6b364 Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36229004

Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 14:43:50 +00:00
kwiberg@webrtc.org
ac2d27d9ae Fix style violations in common_types.h and config.h
Mostly, it's about moving constructors and descructors to the .cc
files, so that they won't be inlined everywhere.

The reason this CL is so big is that a lot of code was using
common_types.h without declaring a dependency on webrtc_common, which
broke the build once common_types.h started to depend on
common_types.cc.

BUG=163
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26089004

Cr-Commit-Position: refs/heads/master@{#8516}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 14:01:28 +00:00
pbos@webrtc.org
891d48393e Wire up target_media_bitrate in VideoSendStream.
Also wires up target_enc_bitrate in WebRtcVideoEngine2.

BUG=1667,1788
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42479004

Cr-Commit-Position: refs/heads/master@{#8515}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8515 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 13:16:17 +00:00
mflodman@webrtc.org
9dd0ebc379 Remove the default RTP module.
This CL removes the default module owned by ViEEncoder, functionality in
the module to register default modules and the final changes in
rtp_rtcp_impl using default/child modules.

BUG=769
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42509004

Cr-Commit-Position: refs/heads/master@{#8514}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8514 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 12:58:24 +00:00
henrik.lundin@webrtc.org
38d9cc51d5 Add back return statement after FATAL()
Some compilers do not accept that non-void functions end with FATAL()
instead of a return statement. This change adds back a few return
statements that were removed in r8463.

BUG=4344
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42519004

Cr-Commit-Position: refs/heads/master@{#8509}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8509 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 09:43:19 +00:00
magjed@webrtc.org
f09e7b8a4f WebRtcVideoFrame: DCHECK exclusive ownership for non-const pixel access
Add some const safety by DCHECK(HasOneRef()) in non-const GetYPlane. This CL also replaces all incorrect non-const calls with const calls for pixel data access in cricket::VideoFrame. It's easy to call the non-const version of e.g. GetYPlane by mistake, even if only const-access is needed. For example:
const scoped_ptr<cricket::VideoFrame> foo;
const uint8_t* y = foo->GetYPlane();
will actually call the non-const version of GetYPlane.

R=mflodman@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39079004

Cr-Commit-Position: refs/heads/master@{#8507}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8507 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 14:50:19 +00:00
mflodman@webrtc.org
96abda0316 Removing FEC functionality from the default RTP module.
This CL removes the last default module methods used from ViEEncoder and
the default module itself will be removed in a separate CL.

BUG=769
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35309004

Cr-Commit-Position: refs/heads/master@{#8505}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8505 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 13:50:51 +00:00
jmarusic@webrtc.org
9b969e167d AudioEncoderCopyRed: CHECK that encode call doesn't fail
Call to AudioEncoder::Encode fails only if fed bad input, so instead of handling failure, we can just CHECK.
There is also no need to handle case where size of encoded data is larger than allowed maximum, so we just CHECK.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42099005

Cr-Commit-Position: refs/heads/master@{#8504}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8504 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 11:53:45 +00:00
andresp@webrtc.org
749c60217d Moved gypi to avoid presubmit warning about '..' when touching the files.
R=kjellander@webrtc.org,mflodman@webrtc.org
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39299004

Cr-Commit-Position: refs/heads/master@{#8503}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8503 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 11:50:44 +00:00
henrik.lundin@webrtc.org
c5558b7021 Remove AudioCodingModule's dependency on the Module interface
BUG=3520
COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42069004

Cr-Commit-Position: refs/heads/master@{#8500}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8500 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 10:37:46 +00:00
henrik.lundin@webrtc.org
af82f75690 Let Add10MsData method do the encoding work as well
This change essentially makes the Process method a no-op. All it does
now is to return a stored value from the last encoding.

The purpose of this change is to forge the Add... and Process methods
into one and the same.

BUG=3520
COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38229004

Cr-Commit-Position: refs/heads/master@{#8499}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8499 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 10:33:42 +00:00