andresp@webrtc.org
b0c8228755
Remove no longer used SkipEncodingUnusedStreams.
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6753 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-22 07:17:17 +00:00
andresp@webrtc.org
5ab7616983
Remove remains of WEBRTC_NO_STL.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6752 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-22 06:48:58 +00:00
buildbot@webrtc.org
fa5fcd671d
(Auto)update libjingle 71599033-> 71605904
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6751 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 21:55:43 +00:00
buildbot@webrtc.org
e69b061926
(Auto)update libjingle 71575585-> 71599033
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6750 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 20:38:58 +00:00
andrew@webrtc.org
ceafa8cce9
MIPS optimizations for ISAC (patch #2 )
...
Implemented functions:
- WebRtcIsacfix_CalculateResidualEnergy
- WebRtcIsacfix_Spec2Time
- WebRtcIsacfix_Time2Spec
- WebRtcIsacfix_HighpassFilterFixDec32
- WebRtcIsacfix_PCorr2Q32
Gain achieved: aprox. further 5% on top of patch#1 on ISAC encoding path.
The optimizations are bit-exact to the C code, with the excception of the
MIPS DSPr2 variant of the WebRtcIsacfix_Time2Spec function (the accuracy of
the WebRtcIsacfix_Time2Spec MIPS DSPr2 variant is same or better than C
variant). Code verification and improvement achieved have been determined
using the iSACFixtest application.
R=andrew@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19749004
Patch from Ljubomir Papuga <lpapuga@mips.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6749 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 16:43:13 +00:00
tommi@webrtc.org
908f57ed94
Disable GetStatsForInvalidTrack while I rewrite it.
...
TBR=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17969005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6748 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 11:44:39 +00:00
tommi@webrtc.org
756b8462eb
Refactor StatsCollector and associated types.
...
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.
R=xians@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=6745
Review URL: https://webrtc-codereview.appspot.com/18819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6747 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 11:24:17 +00:00
tommi@webrtc.org
fd61a1d693
Revert 6745 "Refactor StatsCollector and associated types."
...
Broke build on android.
> Refactor StatsCollector and associated types.
> * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> * Report ids are now const.
> * Copying of data has been greatly reduced.
> * This change includes preparation work for making GetStats fully async.
>
> R=xians@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/18819004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6746 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 11:05:28 +00:00
tommi@webrtc.org
647e05cfcd
Refactor StatsCollector and associated types.
...
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6745 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 10:55:11 +00:00
pbos@webrtc.org
3c10758b3b
Check before send/receive rtp header extensions.
...
BUG=1788
R=pbos@webrtc.org , tommi@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13949004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6744 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-20 15:27:35 +00:00
pbos@webrtc.org
8fdeee6abf
Implement Base::ConstrainNewCodec2.
...
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6743 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-20 14:40:23 +00:00
jiayl@webrtc.org
3edbaaf337
Ignore empty data in DataChannel::Send to match FF's behavior.
...
BUG=crbug/395205
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6742 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 23:57:50 +00:00
buildbot@webrtc.org
99f6308a2d
(Auto)update libjingle 71460499-> 71464449
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6741 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 23:31:30 +00:00
jiayl@webrtc.org
a0b929b63c
Revert "Reland r6707 with the fix for callclient.cc."
...
Breaking pulse build again.
This reverts commit 3e0bb9b5bf7f616000399e24f1d9622ad6b612f9.
TBR=wu@webrtc.org
BUG=3310
Review URL: https://webrtc-codereview.appspot.com/17979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6740 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 22:28:36 +00:00
buildbot@webrtc.org
196ae6d667
(Auto)update libjingle 71456344-> 71456420
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6739 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 21:41:41 +00:00
buildbot@webrtc.org
3dec81a736
(Auto)update libjingle 71456173-> 71456344
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6738 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 21:39:56 +00:00
jiayl@webrtc.org
a6e8cf8fb7
Reland r6707 with the fix for callclient.cc.
...
TBR=mallinath@webrtc.org
BUG=3310
Review URL: https://webrtc-codereview.appspot.com/13039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6737 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 21:34:11 +00:00
minyue@webrtc.org
f563e85ab0
This is to re-open an earlier CL
...
https://webrtc-codereview.appspot.com/16619005/
which is reverted due to an issue in audio conference mixer.
This issue has been solved in
https://webrtc-codereview.appspot.com/20779004/
BUG=webrtc:3155
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18819005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6736 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 21:11:27 +00:00
buildbot@webrtc.org
60e65b11c1
(Auto)update libjingle 71452608-> 71453580
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6735 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 21:07:50 +00:00
jiayl@webrtc.org
8636fc852e
Creates the default track if the remote media content is send-only and there is no stream in the SDP.
...
BUG=2628
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6734 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 20:54:27 +00:00
tkchin@webrtc.org
ff50debd37
Runtime guard for iOS7 property.
...
BUG=3487
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6733 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 17:17:59 +00:00
tkchin@webrtc.org
9343cf67a9
Fix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS.
...
BUG=3581
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6732 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 17:13:28 +00:00
pbos@webrtc.org
ba92c52570
Disable GetStats on DrMemory.
...
Flakes/fails on DrMemory Full just like the implementation in
webrtcvideoengine.cc.
BUG=3482
R=sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6731 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 13:33:48 +00:00
minyue@webrtc.org
026859b983
This is related to an earlier CL of enabling Opus 48 kHz.
...
https://webrtc-codereview.appspot.com/16619005/
It was reverted due to a build bot error, which this CL is to fix. The problem was that when audio conference mixer receives audio frames all at 48 kHz and mixed them, it uses Audio Processing Module (APM) to do a post-processing. However the APM cannot handle 48 kHz input. The current solution is not to allow the mixer to output 48 kHz.
TEST=locally solved https://webrtc-codereview.appspot.com/16619005/
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6730 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 12:28:28 +00:00
pbos@webrtc.org
e6f84ae8a6
Initial WebRtcVideoEngine2::GetStats().
...
Also forward-declaring and moving WebRtcVideoRenderer out of header.
BUG=1788
R=pthatcher@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6729 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 11:11:55 +00:00
pbos@webrtc.org
e9e4253a3c
Sleep in ThreadTest thread functions.
...
Prevents busy loops that really mess up Valgrind's thread scheduling,
this brings runtimes from up to minutes down to milliseconds.
BUG=
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6728 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 10:12:50 +00:00
pbos@webrtc.org
d1ea06b3d5
Restart VideoReceiveStreams in WebRtcVideoEngine2.
...
Puts VideoReceiveStreams in a wrapper, WebRtcVideoReceiveStream that
contain their state (configs). WebRtcVideoRenderer (the wrapper between
webrtc::VideoRenderer and cricket::VideoRenderer) has also been merged
into WebRtcVideoReceiveStream.
Implements and tests setting codecs with new FEC settings as well as RTP
header extensions on already existing receive streams.
BUG=1788
R=pthatcher@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6727 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 09:35:58 +00:00
buildbot@webrtc.org
c31651d847
(Auto)update libjingle 71378257-> 71410012
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 08:22:39 +00:00
kwiberg@webrtc.org
e364ac902f
AudioBuffer: Optimize const accesses to arrays that autoconvert int16<->float
...
Specifically, when someone asks for a const pointer to the int16
version of the array, there's no need to invalidate the float version
of that array, and vice versa. (But obviously, invalidation still has
to happen when someone asks for a non-const pointer.)
R=aluebs@webrtc.org , andrew@webrtc.org , minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6725 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 07:50:29 +00:00
andrew@webrtc.org
c145668dc8
Reduce runtime of RingBufferTest by a factor of 100.
...
This test was needlessly long.
TBR=pbos
Review URL: https://webrtc-codereview.appspot.com/15029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6724 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 23:16:44 +00:00
wu@webrtc.org
4f5da030f1
Use _numMixedParticipants instead of audioFrameList->size() to determine if there're more than one participants.
...
There are two audioFrameLists. The previous check wouldn't work correctly if each list had a single member.
TEST=chrome https://apprtc.appspot.com/?debug=loopback&video=false and verify e2e delay stats
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6723 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 22:19:21 +00:00
mallinath@webrtc.org
aa93611375
Connect to the turn server if address cannot be resolved by the browser by using
...
unresolved address. This case is only considered for TCP sockets. P2P layer will
assume socket will do the resolve by using a proxy.
BUG=3384
R=jiayl@webrtc.org , juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6722 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 21:55:04 +00:00
mallinath@webrtc.org
e5995aadd5
Assigning a priority to TURN server list passed to PeerConnection. First entry in the TURN server list will get the highest priotity and so forth.
...
This priority will be used in calculating the candidate priority generated from the server. This will allow candidate generated from server to have unique priority.
BUG=3223
R=jiayl@webrtc.org , juberti@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6721 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 18:23:52 +00:00
jiayl@webrtc.org
e10d28cf14
fix
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6720 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 17:07:49 +00:00
stefan@webrtc.org
8b94e3da0f
Fix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled.
...
This broke bandwidth estimation for calls without abs-send-time is enabled, but where RTX was.
BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6719 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 16:10:14 +00:00
aluebs@webrtc.org
4065988108
Remove unused ExperimentalNS API in AudioProcessing
...
R=andrew@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6718 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 11:32:09 +00:00
kwiberg@webrtc.org
2b6bc8d84f
AudioBuffer: Eliminate the SplitChannelBuffer class
...
It's just a container for two IFChannelBuffers, and doesn't earn its
keep. The main problem is that the number of methods it needs that
just forward calls to either of its two IFChannelBuffers was already
large, and was about to grow.
R=aluebs@webrtc.org , minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6717 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 09:46:37 +00:00
pbos@webrtc.org
5301b0f1fc
Move additional state into WebRtcVideoSendStream.
...
Prevents having two places where codecs etc. are set up and allows us to
avoid creating the underlying VideoSendStream before send codecs are
set up.
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6716 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 08:51:46 +00:00
aluebs@webrtc.org
2561d52460
Simplify AudioBuffer::mixed_low_pass_data API
...
R=andrew@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6715 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 08:27:39 +00:00
kwiberg@webrtc.org
af93fc08a1
AudioBuffer: Let ChannelBuffer handle bounds checking of channel parameter
...
R=aluebs@webrtc.org , minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6714 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 08:18:33 +00:00
kwiberg@webrtc.org
2ade42bd96
Add unit test for MediaFile WAV file writing
...
R=aluebs@webrtc.org , andrew@webrtc.org , minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6713 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 08:11:32 +00:00
tkchin@webrtc.org
4a472fb18d
Fixes up rtc so that it compiles on iOS 8 SDK.
...
Adds support for UIInterfaceOrientationUnknown (new with in SDK) and makes it the same as
UIInterfaceOrientationPortrait.
R=noahric@google.com , tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13029004
Patch from David Maclachlan <dmaclach@chromium.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6712 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 00:21:59 +00:00
wu@webrtc.org
52eddec71b
Revert 6707 "Add support of multiple STUN servers in UDPPort."
...
Reason:
Breaks the build on callclient.cc.
> Add support of multiple STUN servers in UDPPort.
> Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.
>
> I discovered a bug in SocketAddress while working on this. It didn't consider two addresses unequal if they have unresolved IP and different hosts. It's fixed now.
>
> BUG=3310
> R=mallinath@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/13879004
TBR=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6711 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 00:03:24 +00:00
minyue@webrtc.org
c56ae63ea6
r6709 lacks a change in BUILD.gn
...
BUG=
R=marpan@google.com , marpan@webrtc.org , pbos@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6710 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 22:18:49 +00:00
minyue@webrtc.org
74aaf29a0f
Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.
...
The filter is an exponential filter borrowed from video coding module.
The method is written in a new class called PacketLossProtector (not sure if the name is nice), which can be used in the future for more sophisticated logic.
BUG=
R=henrika@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6709 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 21:28:26 +00:00
wu@webrtc.org
4c3e9917e7
Make sure b lines appear before all the a lines. Per RFC 4566, the order of media description should be:
...
m= (media name and transport address)
i=* (media title)
c=* (connection information -- optional if included at
session level)
b=* (zero or more bandwidth information lines)
k=* (encryption key)
a=* (zero or more media attribute lines)
BUG=2260
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6708 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 21:03:13 +00:00
jiayl@webrtc.org
46fb331bc5
Add support of multiple STUN servers in UDPPort.
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Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.
I discovered a bug in SocketAddress while working on this. It didn't consider two addresses unequal if they have unresolved IP and different hosts. It's fixed now.
BUG=3310
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6707 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 20:55:31 +00:00
tkchin@webrtc.org
2e3c97ddf5
Compile-time guard for iOS7 specific property.
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BUG=3487
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6706 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 19:59:05 +00:00
buildbot@webrtc.org
a8d8ad2be6
(Auto)update libjingle 71240799-> 71250251
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6705 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 14:23:08 +00:00
stefan@webrtc.org
4070b1db53
Print an info log instead of return an error if an external encoder is de-registered, but no corresponding internal encoder can be registered automatically.
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This is not an error case if for instance an external h.264 encoder is registered, but no internal implementation exists.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6704 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 11:20:40 +00:00