298 Commits

Author SHA1 Message Date
henrikg
91d6edef35 Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.

Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1335923002

Cr-Commit-Position: refs/heads/master@{#9964}
2015-09-17 07:24:51 +00:00
henrikg
384194369b Consolidate constructormagic macros with Chromium version and remove Chromium override.
Part of work removing dependency on Chromium's base.

Only adds "= delete". From https://codereview.chromium.org/1151443003 :
"This will guarantee the error to be at compile time, and not rely on the call visibility (private)."

In consequence of that change, fixed an illegal copy and removed a bunch of unused variables.

Depends on https://codereview.webrtc.org/1345433002/

BUG=chromium:468375
(in particular comment #37)
NOTRY=true

Review URL: https://codereview.webrtc.org/1342543004

Cr-Commit-Position: refs/heads/master@{#9954}
2015-09-16 13:33:25 +00:00
henrikg
3c089d751e Add RTC_ prefix to contructormagic macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.

* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS

Related CL: https://codereview.webrtc.org/1335923002/

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1345433002

Cr-Commit-Position: refs/heads/master@{#9953}
2015-09-16 12:37:52 +00:00
Fredrik Solenberg
709ed67c38 Move instantiation of webrtc::Call into a MediaController class so that it can be used for both audio and video media channels.
I'm not super happy with the GetVoE() function added on MediaEngineInterface, but this will eventually be gone, once webrtc::Call owns the shared VoE state (or initially, maps ADM* to an implicitly created VoE).

BUG=webrtc:4690
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1269863005 .

Cr-Commit-Position: refs/heads/master@{#9939}
2015-09-15 10:26:45 +00:00
pbos
1cb121dea4 Reset frame timestamp epoch for new capturers.
Incoming frames usually have an epoch of time since the capturer was
created or similar, not any fixed-time epoch. As such, setting a new
capturer resulted in delivering frames with older timestamps which
caused these frames to be dropped before encoding.

BUG=webrtc:4994
R=stefan@webrtc.org
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1345473002

Cr-Commit-Position: refs/heads/master@{#9934}
2015-09-14 18:38:43 +00:00
solenberg
1dd98f3219 - Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel)
- Rename VideoChannel::MuteStream() -> SetVideoSend() (incl. media channel)
- Collapse NnChannel::SetChannelOptions() into the above.
- Collapse VoiceChannel::SetLocalRenderer into SetAudioSend().

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1311533009

Cr-Commit-Position: refs/heads/master@{#9915}
2015-09-10 08:57:20 +00:00
Magnus Jedvert
f6901b06b8 Remove NullVideoFrame
This class is not used.

R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1309743008 .

Cr-Commit-Position: refs/heads/master@{#9910}
2015-09-09 16:57:48 +00:00
solenberg
66f43392a3 Remove [Voice|Video]MediaChannel::GetOptions().
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1324853003

Cr-Commit-Position: refs/heads/master@{#9904}
2015-09-09 08:36:31 +00:00
solenberg
bb741b3afa Remove GetOutputScaling from VoiceMediaChannel.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1331443003

Cr-Commit-Position: refs/heads/master@{#9870}
2015-09-07 10:56:45 +00:00
stefan
658910cc3c Revert "Speculative revert of "- Move test cases for more natural ordering.""
Did not resolve the build bot issue.

This reverts commit 02d283a6ff5364d94aa88f5f5df4cfd3a5411346.

BUG=
TBR=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1324123002

Cr-Commit-Position: refs/heads/master@{#9849}
2015-09-03 12:48:36 +00:00
Stefan Holmer
02d283a6ff Speculative revert of "- Move test cases for more natural ordering."
This reverts commit c20a5dc9305b988ca173cd63e606124b02e6d54c.

BUG=webrtc:4959
R=solenberg@webrtc.org
TBR=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1309313008 .

Cr-Commit-Position: refs/heads/master@{#9829}
2015-09-01 13:48:31 +00:00
Fredrik Solenberg
c20a5dc930 - Move test cases for more natural ordering.
- Get rid of the CoInitialize tests for WVoE/WViE.

BUG=webrtc:4690
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1319163002 .

Cr-Commit-Position: refs/heads/master@{#9817}
2015-08-31 09:14:05 +00:00
Peter Kasting
dce40cf804 Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 21:52:45 +00:00
Magnus Jedvert
c232096eba Remove cricket::VideoProcessor and AddVideoProcessor() functionality
This functionality is not used internally in WebRTC. Also, it's not safe, because the frame is supposed to be read-only, and it will likely not work for texture frames.

R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1296113002 .

Cr-Commit-Position: refs/heads/master@{#9753}
2015-08-21 09:40:42 +00:00
Peter Thatcher
c2ee2c86f9 Refactor the relationship between BaseChannel and MediaChannel so that we send over all the parameters in one method call rather then having them broken up into multiple method calls. This should allow future refactorings of the WebRtcVideoEngine2 to not recreate configurations so many times, and have more simple code as well.
R=deadbeef@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1229283003 .

Cr-Commit-Position: refs/heads/master@{#9690}
2015-08-07 23:05:42 +00:00
Fredrik Solenberg
bd10ee8bd3 Tiny cleanups.
BUG=webrtc:4690
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1272163002 .

Cr-Commit-Position: refs/heads/master@{#9678}
2015-08-05 10:18:18 +00:00
jbauch
fabe2c961f Remove deprecated functions.
This CL removes some functions that are marked as deprecated. Chromium
has been updated in https://crrev.com/7dee3f68b7699ad72c7fc4d75332f72703313849
to call the new functions.

Review URL: https://codereview.webrtc.org/1237613003

Cr-Commit-Position: refs/heads/master@{#9598}
2015-07-16 20:43:27 +00:00
qiangchen
c27d89fdc6 Let WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame carry the input frame's timestamp to output frame.
Essentially we are carrying over the capture timestamp to the encoded frame sent out, so the frame lengths will contain no noise.

Review URL: https://codereview.webrtc.org/1225153002

Cr-Commit-Position: refs/heads/master@{#9597}
2015-07-16 17:27:23 +00:00
Peter Thatcher
a9b4c32052 Nuke buffered latency mode. It's not actually working, and it's not used. It's just dead code complexity.
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1226093010 .

Cr-Commit-Position: refs/heads/master@{#9593}
2015-07-16 10:47:39 +00:00
Zeke Chin
71f6f4405c iOS HW H264 support.
First step towards supporting H264 on iOS. More tuning/experimentation
required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini.
Future work to get it working on OS/X, simulator (renders black screen
currently) and with the Android AppRTCDemo. Currently protected with a
compile time guard.

BUG=4081
R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1187573004.

Cr-Commit-Position: refs/heads/master@{#9515}
2015-06-29 21:35:08 +00:00
Peter Boström
eb82309d06 Remove FileMediaEngine.
This is currently only used in libjingle/examples/call which is
deprecated and not currently building.

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

Review URL: https://codereview.webrtc.org/1169833004.

Cr-Commit-Position: refs/heads/master@{#9415}
2015-06-11 08:27:36 +00:00
Henrik Lundin
441f634731 Re-land r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
(This reverts commit 3fbf3f8841b5460503fb646eaedcb063620434a8.)

The original submission was reverted because it broke the Chrome build. This is fixed in patch set 2 of this change by keeping the old MediaConstraintsInterface string kExperimentalEchoCancellation. It will be removed once the Chrome code has been updated.

Original description:
"We use this Config struct for enabling/disabling Extended filter mode in AEC. This change renames it to ExtendedFilter for readability reasons. The corresponding media constraint is also renamed to kExtendedFilterEchoCancellation.

The old Config is kept in parallel with the new during a transition period. This is to avoid problems with API breakages. During this period, if any of the two Configs are enabled, the extended filter mode is engaged in APM. That is, the two Configs are combined with an "OR" operation.

This change also renames experimental_aec in AudioOptions to extended_filter_aec."

BUG=webrtc:4696
R=bjornv@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1151573021.

Cr-Commit-Position: refs/heads/master@{#9401}
2015-06-09 14:03:23 +00:00
Henrik Lundin
3fbf3f8841 Revert r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
This reverts commit 5f4b7e2873864c61e2ad6d88679dcd5d321bfd16, since it
broke some of the build bots.

BUG=4696
TBR=bjornv@webrtc.org

Review URL: https://codereview.webrtc.org/1166463006

Cr-Commit-Position: refs/heads/master@{#9380}
2015-06-05 09:04:20 +00:00
Henrik Lundin
5f4b7e2873 Rename APM Config DelayCorrection to ExtendedFilter
We use this Config struct for enabling/disabling Extended filter mode
in AEC. This change renames it to ExtendedFilter for readability
reasons. The corresponding media constraint is also renamed to
kExtendedFilterEchoCancellation.

The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, if any of the two Configs are enabled, the extended filter
mode is engaged in APM. That is, the two Configs are combined with an
"OR" operation.

This change also renames experimental_aec in AudioOptions to extended_filter_aec.

BUG=4696
R=bjornv@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54659004

Cr-Commit-Position: refs/heads/master@{#9378}
2015-06-05 07:55:40 +00:00
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 fd32f35aff8fc28ec084bddc274de284e0422a57.

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 cdb47a4533b7b1e29e803ed6591a68bb1a4f1692.

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 208a2294cde839025318f1b3d57559cb0611a4e7.
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 1b1c15cad16de57053bb6aa8a916079e0534bdae.

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