Commit Graph

274 Commits

Author SHA1 Message Date
Henrik Lundin
8e6fd46cc3 Route time-stretching metrics through libjingle
This change connects currentAccelerateRate and currentPreemptiveRate
in webrtc::NetworkStatistics, through corresponding variables in
VoiceReceiverInfo, to googAccelerateRate and googPreemptiveExpandRate.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50179004

Cr-Commit-Position: refs/heads/master@{#9350}
2015-06-02 07:25:03 +00:00
Henrik Lundin
5263b3c1dd Add options for NetEq fast accelerate mode through libjingle
This CL connects RTCConfiguration::audioJitterBufferFastMode in
PeerConnection.java, through libjingle, down to
NetEq::Config::enable_fast_accelerate in native WebRTC.

When enabled, it will allow NetEq to do faster time-compression when
the buffer level is very high.

BUG=4691
R=henrika@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/55479004

Cr-Commit-Position: refs/heads/master@{#9344}
2015-06-01 08:29:55 +00:00
Jelena Marusic
c28a896a7b VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation
BUG=4690

Changes:
1. In MediaEngineInterface changed CreateChannel() to CreateChannel(const AudioOptions&). Plan is to eventually remove Get/SetAudioOptions and the cousins SetDelayOffset and SetDevices.
2. In ChannelManager changed CreateVoiceChannel(...) to CreateVoiceChannel(..., const AudioOptions&).
3. In ChannelManager removed SetEngineAudioOptions, because it is not used and we want to eventually remove SetAudioOptions.
4. Updated MediaEngineInterface implementations and unit tests accordingly.
5. In WebRtcVoiceEngine changed access of Set/ClearOptionOverrides to protected. These are only used by WebRtcVoiceMediaChannel (now a friend). Plan is to rethink the logic behind option overrides.
6. Cosmetics: replaced NULL with nullptr in touched code

R=solenberg@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/56499004

Cr-Commit-Position: refs/heads/master@{#9330}
2015-05-29 13:05:52 +00:00
Henrik Boström
915df4fc30 CaptureManager: Don't stop a capturer at UnregisterVideoCapturer if it did not start in the first place.
This fixes a bug where, if the VideoCapturer failed to start under certain circumstances, the capture manager would cause a callback saying that the capturer stopped even though it never started in the first place. A VERIFY check in VideoSource::SetState would then cause a crash since the state was set to kEnded when it was already in state kEnded (SetState only allows being called when the state changes).

I only noticed this bug while doing a mistake in a separate CL. Not sure how to reliably reproduce said bug on a working build, but I have previously had camera hardware issues where it couldn't start the camera which resulted in the SetState kEnded -> kEnded crash. Hopefully this will fix that.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51039004

Cr-Commit-Position: refs/heads/master@{#9259}
2015-05-22 07:43:10 +00:00
Fredrik Solenberg
9a416bd14e Get rid of unnecessary Terminate() method and worker_thread_ from WebRtcVideoEngine2
BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51879004

Cr-Commit-Position: refs/heads/master@{#9258}
2015-05-22 07:03:48 +00:00
Guo-wei Shieh
17b889b899 Issue 4366: Adapted frames have wrong width and height and are cropped.
When a frame being stretched, the original rotation information is lost. This is to ensure it's carried over.

Also removed StretchToBuffer function as it's not called and dangerous.

BUG=4366
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51869004

Cr-Commit-Position: refs/heads/master@{#9224}
2015-05-19 19:45:26 +00:00
Fredrik Solenberg
ccb49e79fd Remove Soundclip handling from libjingle.
BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51009004

Cr-Commit-Position: refs/heads/master@{#9216}
2015-05-19 09:37:39 +00:00
Henrik Lundin
64dad838e6 Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
The original change was reverted due to a breakage in the chrome build.
This change includes a fix for this.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49329004

Cr-Commit-Position: refs/heads/master@{#9169}
2015-05-11 10:44:20 +00:00
Henrik Lundin
1f629232d5 Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit fd32f35aff.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/55369004

Cr-Commit-Position: refs/heads/master@{#9165}
2015-05-10 09:06:20 +00:00
Henrik Lundin
fd32f35aff Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit cdb47a4533.

Contains a tentative fix to the chrome build breakage caused by the
original change.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47139004

Cr-Commit-Position: refs/heads/master@{#9164}
2015-05-10 09:03:00 +00:00
Henrik Lundin
cdb47a4533 Revert r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit 208a2294cd.
Breaks the Chrome build.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53399004

Cr-Commit-Position: refs/heads/master@{#9161}
2015-05-08 12:03:46 +00:00
Henrik Lundin
208a2294cd Adding a new constraint to set NetEq buffer capacity from peerconnection
This change makes it possible to set a custom value for the maximum
capacity of the packet buffer in NetEq (the audio jitter buffer). The
default value is 50 packets, but any value can be set with the new
functionality.

R=jmarusic@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50869004

Cr-Commit-Position: refs/heads/master@{#9159}
2015-05-08 10:58:51 +00:00
Fredrik Solenberg
4b60c73e74 Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE.
BUG=4574,3109
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49269004

Cr-Commit-Position: refs/heads/master@{#9150}
2015-05-07 12:07:46 +00:00
Peter Boström
81ea54eaac Remove WebRtcVideoEngine.
Leaves a stub file for talk/media/webrtc/webrtcvideoengine.cc until
build files in Chromium have been modified.

BUG=1695,4566
R=mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48339004

Cr-Commit-Position: refs/heads/master@{#9148}
2015-05-07 09:41:10 +00:00
Shao Changbin
e62202fedf Support handling multiple RTX but only generate SDP with RTX associated with VP8.
This implementation registers RTX-APT map inside RTP sender and receiver.
While it only generates SDP with RTX associated with VP8 to make it
compatible with previous Chrome versions.

Should add following changes after reaches stable,
* Use RTX-APT map for building and restoring RTP packets.
* Add RTX support for RED or VP9 in Video engine.
* Set RTX payload type for RED inside FecConfig in EndToEndTest.

BUG=4024
R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36889004

Cr-Commit-Position: refs/heads/master@{#9040}
2015-04-21 12:25:42 +00:00
Karl Wiberg
9478437fde rtc::Buffer improvements
1. Constructors, SetData(), and AppendData() now accept uint8_t*,
     int8_t*, and char*. Previously, they accepted void*, meaning that
     any kind of pointer was accepted. I think requiring an explicit
     cast in cases where the input array isn't already of a byte-sized
     type is a better compromise between convenience and safety.

  2. data() can now return a uint8_t* instead of a char*, which seems
     more appropriate for a byte array, and is harder to mix up with
     zero-terminated C strings. data<int8_t>() is also available so
     that callers that want that type instead won't have to cast, as
     is data<char>() (which remains the default until all existing
     callers have been fixed).

  3. Constructors, SetData(), and AppendData() now accept arrays
     natively, not just decayed to pointers. The advantage of this is
     that callers don't have to pass the size separately.

  4. There are new constructors that allow setting size and capacity
     without initializing the array. Previously, this had to be done
     separately after construction.

  5. Instead of TransferTo(), Buffer now supports swap(), and move
     construction and assignment, and has a Pass() method that works
     just like std::move(). (The Pass method is modeled after
     scoped_ptr::Pass().)

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

Review URL: https://webrtc-codereview.appspot.com/42989004

Cr-Commit-Position: refs/heads/master@{#9033}
2015-04-20 12:03:00 +00:00
Magnus Jedvert
4b76c02362 Roll chromium_revision 8af41b3..dcb0929 (324854:325030)
This is a major libyuv update (almost 200 revisions):
d204db6..32ad6e0

Relevant changes:
* src/third_party/libyuv: d204db6..32ad6e0
* src/third_party/nss: d1edb68..9506806
Details: 8af41b3..dcb0929/DEPS

Since bayer and Q420 format support have been removed from libyuv, all tests related to those format are removed.

Clang version was not updated in this roll.

R=kjellander@webrtc.org
TBR=tommi

Review URL: https://webrtc-codereview.appspot.com/48989004

Cr-Commit-Position: refs/heads/master@{#9008}
2015-04-15 15:22:19 +00:00
Peter Thatcher
56d50288e0 Remove SignalCaptureStateChange from MediaEngine.
It's no longer used by anything.

R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/48069004

Cr-Commit-Position: refs/heads/master@{#8994}
2015-04-14 00:17:36 +00:00
Peter Thatcher
77f0e3f7b6 Remove GetStartCaptureFormat and some related code.
It is no longer used by anything.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48039004

Cr-Commit-Position: refs/heads/master@{#8990}
2015-04-13 17:44:56 +00:00
Noah Richards
99c2fe5d2b Fix NullVideoEngine's CreateChannel implementation.
BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44149004

Cr-Commit-Position: refs/heads/master@{#8980}
2015-04-10 21:32:42 +00:00
Magnus Jedvert
f6c003eda5 cricket::VideoFrameFactory: Handle if created frame is null
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46869004

Cr-Commit-Position: refs/heads/master@{#8972}
2015-04-10 10:44:51 +00:00
Magnus Jedvert
0184057d54 VideoAdapterTest: Replace FileVideoCapturer with FakeVideoCapturer
The unittests are currently flaky due to the use of FileVideoCapturer.

BUG=4317
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49649004

Cr-Commit-Position: refs/heads/master@{#8969}
2015-04-10 09:18:39 +00:00
Thiago Farina
ef88309a6e Cleanup: Forward declare AudioFrame type in voiceprocess.h
No need to include this header since the API is just taking a pointer to
it.

BUG=1092
TEST=./webrtc/build/gyp_webrtc && ninja -C out/Debug
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44059004

Patch from Thiago Farina <tfarina@chromium.org>.

Cr-Commit-Position: refs/heads/master@{#8928}
2015-04-06 10:36:41 +00:00
Henrik Boström
037bad7497 ~CaptureManager: DCHECK(capture_states_.empty()) instead of CHECK until we fix not empty bug.
BUG=chromium:320200
R=perkj@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49579004

Cr-Commit-Position: refs/heads/master@{#8922}
2015-04-02 10:10:18 +00:00
Guo-wei Shieh
64c1e8cda5 Enable CVO by default through webrtc pipeline.
All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined.

Tests completed:
1. android standalone to android standalone
2. android standalone to chrome (with and without this change)
3. android on chrome

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

Committed: https://crrev.com/1b1c15cad16de57053bb6aa8a916079e0534bdae
Cr-Commit-Position: refs/heads/master@{#8905}

Review URL: https://webrtc-codereview.appspot.com/47399004

Cr-Commit-Position: refs/heads/master@{#8917}
2015-04-01 22:33:15 +00:00
Minyue
31331cfd2d Revert "Enable CVO by default through webrtc pipeline."
This reverts commit 1b1c15cad1.

Due to failure on
http://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/4092
and following builds (the test hangs and never finishes).
R=kjellander@webrtc.org
TBR=guoweis@chromium.org
TESTED=Local revert + execution of libjingle_peerconnection_java_unittest show that this is the culprit.

Review URL: https://webrtc-codereview.appspot.com/47909004

Cr-Commit-Position: refs/heads/master@{#8911}
2015-04-01 14:20:11 +00:00
Guo-wei Shieh
1b1c15cad1 Enable CVO by default through webrtc pipeline.
All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined.

Tests completed:
1. android standalone to android standalone
2. android standalone to chrome (with and without this change)
3. android on chrome

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

Review URL: https://webrtc-codereview.appspot.com/47399004

Cr-Commit-Position: refs/heads/master@{#8905}
2015-04-01 02:42:50 +00:00
Magnus Jedvert
1ecfd55044 videoadapter_unittest.cc: Revert removal of '#if defined(HAVE_WEBRTC_VIDEO)'
This CL reverts some parts of "Delete VideoAdapter::AdaptFrame" https://webrtc-codereview.appspot.com/44769004/.

Reason for revert: Should not touch HAVE_WEBRTC_VIDEO since libjingle_media_unittests does not compile without anyway.

BUG=4317
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48699005

Cr-Commit-Position: refs/heads/master@{#8888}
2015-03-30 09:25:04 +00:00
Peter Boström
8ed6a4bba4 Remove unused non-standard capture stats.
Removes 'googCaptureJitterMs' and 'googCaptureQueueDelayMsPerS' from
talk/. The overuse-detection method used is based on encoding time,
so these stats aren't useful enough to warrant having them showing up in
GetStats().

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

Review URL: https://webrtc-codereview.appspot.com/50469004

Cr-Commit-Position: refs/heads/master@{#8874}
2015-03-27 09:01:11 +00:00
Magnus Jedvert
3954e1dfe1 Remove unused implementations in cricket::VideoFrame
This CL moves dummy implementations from cricket::VideoFrame to NullVideoFrame instead.

R=guoweis@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50409004

Cr-Commit-Position: refs/heads/master@{#8873}
2015-03-27 08:48:45 +00:00
Minyue Li
7100dcd317 Adding "usedtx" as Opus codec parameter.
This is according to https://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03

Specifically,

usedtx: specifies if the decoder prefers the use of DTX. values are 1 and 0. If no value is specified, usedtx is assumed to be 0.

BUG=1014
R=juberti@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48499004

Cr-Commit-Position: refs/heads/master@{#8872}
2015-03-27 04:06:35 +00:00
Brave Yao
5225dd8180 If audio ptime is negotiated in SDP, then we would set the audio codec with negotiated packet size if it's allowed. If the negotiated packet size is not supported by the working codec, then we would use the next smallest size.
BUG=4289
TEST=Manual/Auto Test
R=juberti@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44629004

Cr-Commit-Position: refs/heads/master@{#8863}
2015-03-25 23:39:33 +00:00
Bjorn Volcker
bf395c1fc0 Add WebRTC Media Constraint to force using Delay Agnostic AEC on Android
If built-in Echo Cancellation is available on a device it is automatically enabled. The reason is that it in most cases performs better than the WebRTC software echo control for mobile. The drawback is that we can not develop, test and rollout the delay agnostic AEC (DA-AEC) on Android as for desktops.

This CL includes
- adding a media constraint to enable/disable DA-AEC.
- automatically turning on echo cancellation if DA-AEC is enabled.
- a fix in the AEC that enables delay estimation when DA-AEC is enabled, but delay metrics is disabled.
- sets the Config struct ReportedDelay, which controls DA-AEC internally in the AEC.

The test code to verify that it works in AppRTCDemo can be found here:
https://webrtc-codereview.appspot.com/50479004/

BUG=4472
TESTED=locally on N7, N6, Android One
R=glaznev@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48699004

Cr-Commit-Position: refs/heads/master@{#8861}
2015-03-25 21:46:10 +00:00
Magnus Jedvert
e61c64dbb1 Delete NullVideoRenderer
NullVideoRenderer is not used.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51419004

Cr-Commit-Position: refs/heads/master@{#8850}
2015-03-24 15:11:24 +00:00
Magnus Jedvert
ac27e20477 Delete VideoAdapter::AdaptFrame
This CL deletes VideoAdapter::AdaptFrame and replaces the remaining calls with AdaptFrameResolution instead.

I do not expect this CL to fix the flaky VideoAdapterTests yet. I intend to replace FileVideoCapturer with a deterministic FakeVideoCapturer in a follow-up CL.

BUG=4317
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44769004

Cr-Commit-Position: refs/heads/master@{#8848}
2015-03-24 14:18:52 +00:00
kwiberg@webrtc.org
eebcab5ce9 rtc::Buffer: Rename length to size, for conformance with the STL
And add a constructor for creating an uninitialized Buffer of a
specified size.

(I intend to follow up with more Buffer changes, but since it's rather
widely used, the rename is quite noisy and works better as a separate
CL.)

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48579004

Cr-Commit-Position: refs/heads/master@{#8841}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8841 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-24 09:20:19 +00:00
magjed@webrtc.org
14ee8cc9c7 WebRtcVideoFrame: Support odd resolutions
We currently truncate the resolution of frames to a multiple of 4. This is unnecessary as everything supports odd resolutions now.

R=fbarchard@google.com, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43819004

Cr-Commit-Position: refs/heads/master@{#8774}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8774 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 09:22:19 +00:00
perkj@webrtc.org
a78a94e838 Fix RateTracker to set an initial reference time when first updated.
BUG=4442
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43829004

Cr-Commit-Position: refs/heads/master@{#8751}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8751 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 12:45:41 +00:00
changbin.shao@webrtc.org
2d25b44f47 Check associated payload type when negotiate RTX codecs.
At the moment, only payload name is checked when match two RTX codecs.
This will cause wrong behavior of codec negotiation if multiple RTX codecs
are added.

BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34189004

Cr-Commit-Position: refs/heads/master@{#8727}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8727 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 04:15:23 +00:00
guoweis@webrtc.org
00c509ad1c Add concept of whether video renderer supports rotation.
Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation.

Tested with peerconnection_client on windows, AppRTCDemo on Mac.

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

Committed: https://code.google.com/p/webrtc/source/detail?r=8660

Committed: https://code.google.com/p/webrtc/source/detail?r=8661

Review URL: https://webrtc-codereview.appspot.com/43569004

Cr-Commit-Position: refs/heads/master@{#8705}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8705 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 21:38:19 +00:00
guoweis@webrtc.org
fdd1057949 Add CVO support to Vie layer.
1. standard plumbing CVO through vie layer.
2. added a rtp_cvo.h which has both conversion functions from rtp header byte to/from VideoRotation.

WebRTCVideoEngine will later pass the rotation info in SendFrame() through VieVideoFrameI420.

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

Review URL: https://webrtc-codereview.appspot.com/46429007

Cr-Commit-Position: refs/heads/master@{#8703}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8703 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 20:51:50 +00:00
magjed@webrtc.org
afdd5dd372 Revert "Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame""
This reverts r8683 and is a reland of r8682.

Reason for revert: The thread checker in Chromium that crashed has been fixed now.

BUG=1128
TBR=tommi,pbos,pthatcher

Review URL: https://webrtc-codereview.appspot.com/40319004

Cr-Commit-Position: refs/heads/master@{#8696}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8696 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 13:11:43 +00:00
magjed@webrtc.org
b218ff5531 Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame"
This reverts r8682.

Reason for revert: Fails on Chromium FYI content_browsertests

BUG=1128
TBR=tommi,pbos,pthatcher

Review URL: https://webrtc-codereview.appspot.com/47529004

Cr-Commit-Position: refs/heads/master@{#8683}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8683 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-11 15:29:59 +00:00
magjed@webrtc.org
370a72cc3f Remove frame copy from cricket::VideoFrame to I420VideoFrame
BUG=1128
R=pbos@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42249004

Cr-Commit-Position: refs/heads/master@{#8682}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8682 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-11 14:16:16 +00:00
guoweis@webrtc.org
f9a75d99b9 Revert "Add concept of whether video renderer supports rotation."
This reverts commit 0ad48935fc.

TBR=guoweis@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/41199004

Cr-Commit-Position: refs/heads/master@{#8663}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8663 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-10 06:37:41 +00:00
guoweis@webrtc.org
60a2aa0652 Revert "Add concept of whether video renderer supports rotation."
This reverts commit 31d16467ac.

TBR=guoweis@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/47489004

Cr-Commit-Position: refs/heads/master@{#8662}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8662 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-10 06:20:18 +00:00
guoweis@webrtc.org
31d16467ac Add concept of whether video renderer supports rotation.
Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation.

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

Committed: https://code.google.com/p/webrtc/source/detail?r=8660

Review URL: https://webrtc-codereview.appspot.com/43569004

Cr-Commit-Position: refs/heads/master@{#8661}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8661 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-10 06:11:44 +00:00
guoweis@webrtc.org
0ad48935fc Add concept of whether video renderer supports rotation.
Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation.

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

Review URL: https://webrtc-codereview.appspot.com/43569004

Cr-Commit-Position: refs/heads/master@{#8660}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8660 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-10 02:43:40 +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
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