824 Commits

Author SHA1 Message Date
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
pbos@webrtc.org
058b1f17ac Remove GetReceiveBandwidthEstimatorStats.
Removes unnecessary non-standard stats that we don't really make use of.

BUG=
R=pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47379004

Cr-Commit-Position: refs/heads/master@{#8588}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8588 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 08:55:16 +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
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
tommi@webrtc.org
e1b84a0b2b Fix a race reported by tsan.
TSAN complains about this variable not having synchronized access, so
I'm using atomic operations on it instead.
There's no functional difference really though.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42539004

Cr-Commit-Position: refs/heads/master@{#8543}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8543 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-01 08:29:44 +00:00
tommi@webrtc.org
fbef5c6293 Remove lock in ViEFrameProviderBase::IsFrameCallbackRegistered.
After looking at how this class is used I discovered that the threading model isn't completely crazy and that we can remove a lock from a method that's causing the most contention in getStats right now (and probably many other places).  IsFrameCallbackRegistered() looks innocent enough, but it contented with the frame encoder, which can make it wait for tens of milliseconds, unnecessarily.

In addition to removing the lock, I:

* Documented the threading model
* Added checks to guard against regressions
* Reduced lock scope in the Deregister function (not calling out to the notification while holding the lock).
*  made virtual functions non-virtual (since they're not really being overriden) and removed unnecessary method that, if called by other threads than the ctor thread, returned a value that wasn't necessarily correct.

BUG=2822
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40859004

Cr-Commit-Position: refs/heads/master@{#8531}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8531 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-27 15:43:23 +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
pbos@webrtc.org
6c2e506cf4 Workaround Mac align bug for observer_ and crit_.
Unblocks rolling WebRTC into Chromium so we can debug what's actually
going on.

BUG=
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39329004

Cr-Commit-Position: refs/heads/master@{#8528}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8528 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-27 14:28:26 +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
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
pbos@webrtc.org
3e6e271ec3 Implement CpuOveruseMetrics as callbacks.
Adds avg_encode_ms and encode_usage_percent in WebRtcVideoEngine2 and
corresponding stats to VideoSendStream::Stats.

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

Review URL: https://webrtc-codereview.appspot.com/42429004

Cr-Commit-Position: refs/heads/master@{#8513}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8513 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 12:20:24 +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
pkasting@chromium.org
d324546ced Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :
* Move constants into the files/functions that use them
* Declare variables in the narrowest scope possible
* Use correct (expected, actual) order for gtest macros
* Remove unused functions
* Untabify
* 80-column limit
* Avoid C-style casts
* Prefer true typed constants to "enum hack" constants
* Print size_t using the right format macro
* Shorten and simplify code
* Other random cleanup bits and style fixes

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

Review URL: https://webrtc-codereview.appspot.com/36179004

Cr-Commit-Position: refs/heads/master@{#8467}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8467 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 21:29:45 +00:00
mflodman@webrtc.org
50e28166af Move SetTargetSendBitrates logic from default module to payload router.
This cl just moves the logic form the default module
SetTargetSendBitrates to PayloadRouter. There might be glitch / mismatch
in size between trate the vector and rtp modules. This was the same in
the default module and is quite hard to protect from before we have the
new video API.

I also removed some test form rtp_rtcp_impl_unittest that were affected
by this change. The test tests code that isn't implemented, hence the
DISABLED_, and this will never be implemented in the RTP module, rather
the payload router in the future.

BUG=769
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42419004

Cr-Commit-Position: refs/heads/master@{#8453}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8453 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 07:45:45 +00:00
mflodman@webrtc.org
7ac374abd7 Fix shutdown race for ViEEncoder when there is a frame in the encoder.
There is a potential race when deleting a channel and there is a frame
in the encoder. ViEEncoder::SendData can be called after
ViEEncoder::StopThreadsAndRemovePayloadRouter and payload_router is
then already removed.

Until we have the new API in place, use scoped_refptr in ViEChannel and
ViEEncoder and deregister channel/encoder before deleting.

BUG=769
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42019004

Cr-Commit-Position: refs/heads/master@{#8443}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8443 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-20 12:46:21 +00:00
tommi@webrtc.org
e07710cc91 Make SendCodec() lock-free.
Fetching the current codec for sake of gathering stats, is frequently blocked since it's done by acquiring the same lock as is held while encoding frames.  This can mean tens of milliseconds.

To improve this, I'm taking advantage of the fact that the codec information is set on the same thread as is used to query the information.  This means that locking isn't needed for querying this information.  I'm adding checks to make sure debug builds will crash if this isn't followed.

An alternative to this approach could be to add one more lock that is specifically used for the codec information variable.  This would also decouple querying codec information from the encoder itself, but still requires a lock.

This patch depends on making ThreadChecker part of rtc_base_approved:
https://webrtc-codereview.appspot.com/40539004/

BUG=2822
R=mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37779004

Cr-Commit-Position: refs/heads/master@{#8435}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8435 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 17:43:45 +00:00
magjed@webrtc.org
be29b3b4c6 I420VideoFrame: Remove functions set_width, set_height, and ResetSize
The functions set_width, set_height, and ResetSize in I420VideoFrame are not needed and just add complexity.

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

Review URL: https://webrtc-codereview.appspot.com/39939004

Cr-Commit-Position: refs/heads/master@{#8434}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8434 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 15:35:50 +00:00
pbos@webrtc.org
1d0fa5d352 Add RtcpPacketTypeCounter stats to new API.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788

Review URL: https://webrtc-codereview.appspot.com/37489004

Cr-Commit-Position: refs/heads/master@{#8429}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8429 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 12:47:45 +00:00
mflodman@webrtc.org
47d657b68e Remove Set/Get sending status from the default RTP module.
This is now taken care of by the payload router and the calls to set_active.

BUG=769
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42379004

Cr-Commit-Position: refs/heads/master@{#8427}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8427 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 10:30:19 +00:00
magjed@webrtc.org
32c784c266 ViEExternalRendererImpl: Remove dependency to webrtc::VideoFrame
I had to use std::vector, because rtc::Buffer wasn't in rtc_base_approved.

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

Review URL: https://webrtc-codereview.appspot.com/34249004

Cr-Commit-Position: refs/heads/master@{#8426}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8426 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 10:04:17 +00:00
magjed@webrtc.org
f68e186de3 Remove EnableMirroring and MirrorRenderStream
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35239004

Cr-Commit-Position: refs/heads/master@{#8409}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8409 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-18 12:55:17 +00:00
jansson@webrtc.org
97aaf68fed Bump to version 42.
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40739004

Cr-Commit-Position: refs/heads/master@{#8401}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8401 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-18 08:20:23 +00:00
mflodman@webrtc.org
0abc6011b9 Remove SetCaptureDelay from the RTP module.
This is a small step in getting rid of the default module, but also to
eventually delete FrameProviderBase completely.

BUG=769
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34229004

Cr-Commit-Position: refs/heads/master@{#8396}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8396 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-17 16:36:48 +00:00
magjed@webrtc.org
7a91acb94a ViECapturer: Remove unimplemented function declaration |DeliverCodedFrame|
The end goal except cleanup is to remove webrtc::VideoFrame.

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

Review URL: https://webrtc-codereview.appspot.com/35129004

Cr-Commit-Position: refs/heads/master@{#8392}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8392 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-17 14:57:22 +00:00
pbos@webrtc.org
4dd40d6b88 Signal threads for faster receiver destruction.
Unblocks pending threads (render thread + decoder thread) when
destroying renderers and shutting down decoders.

Speeds up SetLocalDescription significantly (10x or so) under
WebRtcVideoEngine2 but also shutdown times in ~ViEChannel and
~ViEReceiver in general.

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

Review URL: https://webrtc-codereview.appspot.com/41959004

Cr-Commit-Position: refs/heads/master@{#8387}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8387 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-17 13:23:27 +00:00
mflodman@webrtc.org
290cb56dca Remove TimeToSendPacket and TimeToSendPadding from the default module.
Thie CL moves the default RTP module logic for TimeToSendPacket and
TimeToSendPadding to PayloadRouter class and asserts on usage of the
default module.

BUG=769
TEST=New unittest.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33319004

Cr-Commit-Position: refs/heads/master@{#8383}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8383 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-17 10:15:47 +00:00
guoweis@webrtc.org
5a7dc39277 This is a code clean up. No functional change intended.
Consolidate the enum for capturer/frame rotation we use through out the code base.

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

Review URL: https://webrtc-codereview.appspot.com/39859004

Cr-Commit-Position: refs/heads/master@{#8365}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8365 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-13 14:32:13 +00:00
mflodman@webrtc.org
2bd299a172 Remove call to RtpRtcp::RegisterSendPayload for the default RTP module.
The send payload type is only checked in RTPSender::CheckPayloadType,
which in turn is only called from SendOutgoingData and never from the
default module anylonger.

BUG=769
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39949004

Cr-Commit-Position: refs/heads/master@{#8357}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8357 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-13 09:52:17 +00:00
pbos@webrtc.org
40367f984b Remove default video encoders for new video API.
Reduces stream creation time significantly. As a side effect also
removes default encoders for receive-only channels.

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

Review URL: https://webrtc-codereview.appspot.com/37049004

Cr-Commit-Position: refs/heads/master@{#8356}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8356 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-13 08:00:42 +00:00
solenberg@webrtc.org
aafbec15f9 Remove ViENetwork::SetBandwidthEstimationConfig() interface since dynamically changing BWE settings isn't necessary now that AIMD is the default.
BUG=3735
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39919005

Cr-Commit-Position: refs/heads/master@{#8351}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8351 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-12 13:21:27 +00:00
pbos@webrtc.org
9e4e524f38 Use an external-only VideoRenderModule in Call.
The default render module instantiated from inside VideoEngine if none
exists instantiates platform-specific code. Call only uses external
rendering, so this is an unneccessary overhead.

BUG=1667
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39069004

Cr-Commit-Position: refs/heads/master@{#8346}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8346 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-12 10:48:55 +00:00
mflodman@webrtc.org
a4ef2ce29d Remove getting max payload length from default module.
Moving functionality to get max payload length from default RTP module
to the payload router.

I'll make a follow up CL changing asserts to DCHECK in rtp_rtcp_impl.cc.

BUG=769
TEST=New unittest and existing sender mtu test
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36119004

Cr-Commit-Position: refs/heads/master@{#8345}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8345 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-12 09:55:05 +00:00
mflodman@webrtc.org
a98e796615 Remove default RTP module functionality for setting CSRC.
ViECapturer is always calling DeliverFrame with an empty CSRC vector, so
this is basically a dead path already. I added a DCHECK in ViEEncoder to
verify this is true.

BUG=769
TEST=Manually verified in Chromium.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39059004

Cr-Commit-Position: refs/heads/master@{#8335}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8335 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-11 15:46:20 +00:00
kjellander@webrtc.org
f58fe0ab2b Rename GYP and GN targets for video capture+render.
This CL performs the following renames of targets to
make GYP and GN more unified and make the targets that
have the same name as the module and include the external
render/capture implementation (the internal one is only
used by WebRTC tests).
This makes it natural to declare dependencies in GN
without having to specify the target.

Summary of the renames:
GYP:
video_render_module_impl -> video_render (new target)
video_capture_module_impl -> video_capture (new target)

GN:
video_capture -> video_capture_module (now identical to the GYP target)
video_capture_impl -> video_capture

video_render -> video_render_module (now identical to the GYP target)
video_render_impl -> video_render

BUG=456815
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35099004

Cr-Commit-Position: refs/heads/master@{#8323}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8323 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-11 07:47:47 +00:00
pbos@webrtc.org
f4c10d24dc Always use DeliverI420Frame in WebRtcVideoEngine.
Moves native_handle() path to DeliverI420Frame and CHECKs that
DeliverFrame is not being used anymore.

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

Review URL: https://webrtc-codereview.appspot.com/38019004

Cr-Commit-Position: refs/heads/master@{#8312}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8312 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-10 10:20:38 +00:00
mflodman@webrtc.org
948d61724c Create a separate thread for pacing.
This CL moves the pacer out from the regular module process thread to
instead use one thread per pacer. This is to get better accuracy for the
paced packets and to avoid overusing the module process thread.

BUG=
TEST=existing tests
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41839004

Cr-Commit-Position: refs/heads/master@{#8308}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8308 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-10 08:59:00 +00:00
tommi@webrtc.org
8e612aba60 Remove voice_engine_ member variable and GetVoiceEngine() from ViEChannelManager.
This is dead code right now and since the implementation of GetVoiceEngine() grabbed a lock and returned a raw pointer, it's not to be trusted anyway :)

R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/37869005

Cr-Commit-Position: refs/heads/master@{#8306}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8306 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-10 08:16:02 +00:00
pbos@webrtc.org
0d852d5c27 Use VideoReceiveStream as an ExternalRenderer.
Removes AddRenderCallback from ViERenderer and implements
VideoReceiveStream on top of DeliverI420Frame like WebRtcVideoEngine
currently does today.

Also adds ::IsTextureSupported() to the VideoRenderer interface to
permit querying whether an external renderer supports texture rendering.

R=stefan@webrtc.org
TBR=mflodman@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/34169004

Cr-Commit-Position: refs/heads/master@{#8299}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8299 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-09 15:15:24 +00:00
changbin.shao@webrtc.org
f31f56d8d4 Remove default arguments in EncodedImageCallback.
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39719004

Cr-Commit-Position: refs/heads/master@{#8289}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8289 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-09 09:14:48 +00:00
tommi@webrtc.org
7a57f8f101 Reland 8203 "Reducing locking in OveruseFrameDetect..."
The issue that was causing the thread checker to report error, turned out to be unrelated.

> Revert 8203 "Reducing locking in OveruseFrameDetector and increa..."
>
> Broke tests in Chrome for some reason:
>
> [ RUN      ] WebRtcAecDumpBrowserTest.CallWithAecDump
> [80131:1287:0129/074432:30561723987517:ERROR:vt_video_decode_accelerator.cc(132)] Failed to create VTDecompressionSession: codecOpenErr (-8973)
> [80129:1287:0129/074432:30562276677373:INFO:CONSOLE(64)] "Looking at video in element remote-view-1", source: http://127.0.0.1:61401/media/webrtc_test_utilities.js (64)
> [80129:1287:0129/074432:30562281435788:INFO:CONSOLE(64)] "Looking at video in element remote-view-2", source: http://127.0.0.1:61401/media/webrtc_test_utilities.js (64)
> [80129:1287:0129/074432:30562315329399:INFO:CONSOLE(800)] "Negotiating call...", source: http://127.0.0.1:61401/media/peerconnection-call.html (800)
> [80133:29187:0129/074432:30562402039578:FATAL:overuse_frame_detector.cc(388)] Check failed: processing_thread_.CalledOnValidThread().
> 0   libbase.dylib                       0x000000010dfd688f base::debug::StackTrace::StackTrace() + 47
> 1   libbase.dylib                       0x000000010dfd68e3 base::debug::StackTrace::StackTrace() + 35
> 2   libbase.dylib                       0x000000010e030076 logging::LogMessage::~LogMessage() + 70
> 3   libbase.dylib                       0x000000010e02f0c3 logging::LogMessage::~LogMessage() + 35
> 4   libcontent.dylib                    0x000000011d8c0cd5 webrtc::OveruseFrameDetector::TimeUntilNextProcess() + 245
> 5   libcontent.dylib                    0x000000011d31ddfd webrtc::ProcessThreadImpl::Process() + 525
> 6   libcontent.dylib                    0x000000011d31d836 webrtc::ProcessThreadImpl::Run(void*) + 38
> 7   libcontent.dylib                    0x000000011d10c390 webrtc::ThreadPosix::Run() + 288
> 8   libcontent.dylib                    0x000000011d10c076 webrtc::StartThread(void*) + 38
> 9   libsystem_pthread.dylib             0x00007fff8e667899 _pthread_body + 138
> 10  libsystem_pthread.dylib             0x00007fff8e66772a _pthread_struct_init + 0
> 11  libsystem_pthread.dylib             0x00007fff8e66bfc9 thread_start + 13
>
>
> > Reducing locking in OveruseFrameDetector and increasing constness.
> >
> > I also added a few TODOs there to see what we can do to reduce the chance of contention.
> > To catch regressions, I've started using the ThreadChecker class on the processing thread but it might also be a good idea to add similar checks for other known threads such as the thread we receive frames on.  I'm sure we can reduce locking even further.
> >
> > BUG=2822
> > R=asapersson@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/33129004
>
> TBR=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/34079004

TBR=tommi@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/35029004

Cr-Commit-Position: refs/heads/master@{#8287}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8287 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-08 18:29:12 +00:00
mflodman@webrtc.org
02270cd718 Implementing a packet router class, used to route RTP packets to the
sending RTP module for the specified simulcast layer a frame belongs to.
This CL also removes the corresponding functionality from the RTP RTCP
module and fixes lint warnings in the files touched.

BUG=769
TEST=New unittest and manual tests
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39629004

Cr-Commit-Position: refs/heads/master@{#8267}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8267 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-06 13:10:39 +00:00
tommi@webrtc.org
0c3e12b7bf Revamp the ProcessThreadImpl implementation.
* Add a new WakeUp method that gives a module a chance to be called back right away on the worker thread.
* Wrote unit tests for the class.
* Significantly reduce the amount of locking.
  - ProcessThreadImpl itself does a lot less locking.
  - Reimplemented the way we keep track of when to make calls to Process.
    This reduces the amount of calls to TimeUntilNextProcess and since most implementations of that function grab a lock, this means less locking.
* Renamed ProcessThread::CreateProcessThread to ProcessThread::Create.
* Added thread checks for Start/Stop.  Threading model of other functions is now documented.
* We now log an error if an implementation of TimeUntilNextProcess returns a negative value (some implementations do, but the method should only return a positive nr of ms).
* Removed the DestroyProcessThread method and instead force callers to use scoped_ptr<> to maintain object lifetime.

BUG=2822
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35999004

Cr-Commit-Position: refs/heads/master@{#8261}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8261 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-06 09:44:45 +00:00
tommi@webrtc.org
875c97ed9d Remove SetNotAlive method from the thread class.
Also cleaning up methods with the same name in other classes that are derived from the above method.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41759004

Cr-Commit-Position: refs/heads/master@{#8242}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8242 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-04 11:12:39 +00:00
asapersson@webrtc.org
4414939954 Add method for incrementing RtpPacketCounter. Removes duplicate code.
Correction to check if rtx is enabled on send-side (and not receive) when updating rtx send bitrate stat.

Remove unneeded guarded by annotations.

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

Review URL: https://webrtc-codereview.appspot.com/41729004

Cr-Commit-Position: refs/heads/master@{#8239}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8239 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-04 08:35:21 +00:00
pkasting@chromium.org
19f3f71c98 Fix apparent typo: int -> char.
The surrounding similar methods all used unsigned char, using unsigned int in
this case looks like an accident, especially since the function passes on the
value in question to a function expecting a uint8.

BUG=none
TEST=none
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40529004

Cr-Commit-Position: refs/heads/master@{#8228}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8228 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-02 19:44:42 +00:00
pkasting@chromium.org
026b892e72 Using << on an int8_t or uint8_t will output a character rather than a number.
Places that do this need to cast to int to get the desired behavior.

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

Review URL: https://webrtc-codereview.appspot.com/40579004

Cr-Commit-Position: refs/heads/master@{#8223}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8223 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 19:54:19 +00:00