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Author SHA1 Message Date
fbarchard@google.com
30cd5b5278 libyuv roll to r986 for c89 fix to cpu_id. 
BUG=none
TESTED=cl cpu_id.cc
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-31 17:28:46 +00:00
solenberg@webrtc.org
caeae4680c Add tests for the RBE RemoveStream() API. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5812 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-31 13:33:39 +00:00
henrik.lundin@webrtc.org
d0a81d91ff VoE Channel: Don't register codecs when stopping receiver 
VoiceEngine's Channel::StopReceiving() would call
RegisterReceiveCodecsToRTPModule(), which caused some errors
with RED and ULP-FEC. In particular, an error message would be
printed when hanging up a call in voe_cmd_test application.

BUG=3085
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5811 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-31 07:31:53 +00:00
fischman@webrtc.org
fe16488184 AppRTCDemo(android): specify DtlsSrtpKeyAgreement:true in CreatePeerConnection's constraints. 
This is required to interop with Chrome now that SDES is disabled in Chrome (as of r5640).

BUG=2774
R=jiayl@chromium.org

Review URL: https://webrtc-codereview.appspot.com/10749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5809 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-28 19:58:03 +00:00
fischman@webrtc.org
4f2bd68744 Silence pointless LS_WARNING about port 0 for active-only candidates. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5808 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-28 18:13:34 +00:00
wu@webrtc.org
987f2c9aae (Auto)update libjingle 63913264-> 63948945 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5807 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-28 16:22:19 +00:00
kjellander@webrtc.org
0aa04f9f24 Restore support for code coverage in WebRTC 
In https://codereview.chromium.org/68193002
Chromium dropped the support for the coverage=1 flag.
This restores it for WebRTC purposes for the Linux platform.

TEST=Manually ran the coverage steps on my machine, verified
that .gcno files are generated.

BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-28 13:14:00 +00:00
wu@webrtc.org
f7d501d48a (Auto)update libjingle 63884381-> 63913264 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 23:48:25 +00:00
andrew@webrtc.org
a5586b50e5 Protect ENABLE_PROFILING to fix profiling=1. 
Chromium defines ENABLE_PROFILING under the gyp flag profiling=1. This
corrects the resulting mulitple defintion error:
../../talk/base/profiler.h:61:9: error: 'ENABLE_PROFILING' macro redefined [-Werror]
#define ENABLE_PROFILING

and allows us to use profiling=1 in standalone builds.

TESTED=build passes with profiling=1
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5804 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 22:44:13 +00:00
fischman@webrtc.org
dd0b99debb Roll libvpx 258445:259973. 
- 259973: unbreak iOS simulator build (-mssse3)
- 259953: add a missing file (follow-up to r259946)
- 259946: Disable assembly optimizations in MemorySanitizer builds.
- 259324: disable function level linking when building vp8_asm_enc_offsets.c

BUG=3126
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10829005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5803 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 22:27:54 +00:00
andrew@webrtc.org
fff3fd35a6 Add arm64 to typedefs.h 
This is the first step to get a buildable chrome_shell_apk for arm64.

BUG=chromium:354405
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10819004

Patch from Primiano Tucci <primiano@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 19:48:53 +00:00
andresp@webrtc.org
5a0218c794 Allow loopback tests to do TURN when served from webrtc.googlecode.com. 
BUG=3037
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5801 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 19:24:45 +00:00
wu@webrtc.org
cfe5e9c894 (Auto)update libjingle 63837929-> 63884381 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5800 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 17:03:58 +00:00
andresp@webrtc.org
6b17be0bf8 Add svn mime-type properties to loopback_test files so they can be served from: 
https://webrtc.googlecode.com/svn/trunk/webrtc/tools/loopback_test/loopback_test.html


git-svn-id: http://webrtc.googlecode.com/svn/trunk@5799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 10:52:09 +00:00
andrew@webrtc.org
b13a7d5b1c Don't disable experimental AGC in audioproc. 
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5798 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 00:11:11 +00:00
henrike@webrtc.org
b0ecc1c6fb (Auto)update libjingle 63777286-> 63837929 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5797 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 22:44:28 +00:00
andrew@webrtc.org
b6dfbed1dc Exclude TwoStreamsSendAndFailUnsignalledRecvInOneToOne from TSAN. 
Example failure:
http://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan/builds/1458

TBR=wu@webrtc.org
BUG=2380

Review URL: https://webrtc-codereview.appspot.com/10759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5796 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 22:22:46 +00:00
fischman@webrtc.org
b25576a75b talk/: enable _DEBUG in Debug for all posix 
Chromium's build/common.gypi defines _DEBUG for Debug builds _except_ on
(OS=="mac" OS=="ios").  But libjingle uses _DEBUG on all platforms so define it on all posix (chromium covers non-posix separately and fine).

BUG=webrtc:3101
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/10699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5795 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 21:53:47 +00:00
andresp@webrtc.org
44caf01c34 Re-submit: rev5775 
Modify bitrate controller to update bitrate based on process call and not
only whenever a RTCP receiver block is received.

Additionally:
 Add condition to only start rampup after a receiver block is received. This was same as old behaviour but now an explicit check is needed to verify process does not ramps up before the first block.

 Fix logic around capping max bitrate increase at 8% per second. Before it was only increasing once every 1 second and each increase would be as high as 8%. If receiver blocks had a different interval before it would lose an update or waste an update slot and not ramp up as much as a 8% (e.g. if RTCP received < 1 second).

 Did not touch decrease logic, however since it can be triggered more often it
 may decrease much faster and closer to the original written cap of once every
 300ms + rtt.

Note:
 rampup_tests.cc don't seem to be affected by this since there is no packet loss or REMB that go higher than expected cap.
 bitrate_controller_unittests.cc are don't really simulate a clock and the process thread, but trigger update by inserting an rtcp block.

BUG=3065
R=stefan@webrtc.org, mflodman@webrtc.org

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5794 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 21:00:21 +00:00
henrike@webrtc.org
1ca08f65e3 Fix after auto update in r5787. APPRTCVideoView.h/m was removed incorrectly. 
BUG=3121
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5793 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 16:42:14 +00:00
jiayl@webrtc.org
7ee0c16edd Makes ScreenCapturerMac exclude the window specified in DesktopCapturer::SetExcludedWindow. 
No behavior change for now since Chromium has not been updated to call SetExcludedWindow.

BUG=2789
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/10299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5792 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 15:57:43 +00:00
solenberg@webrtc.org
4e65602886 Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5791 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 14:32:47 +00:00
andresp@webrtc.org
d09d074827 Protect write of send_target_bitrate. 
This issue was catch by tsan bot.

BUG=3065
R=stefan@webrtc.org, andrew

Review URL: https://webrtc-codereview.appspot.com/10619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5790 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 14:27:34 +00:00
henrike@webrtc.org
5fb7428496 (Auto)update libjingle 63775799-> 63776369 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5789 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 02:00:10 +00:00
henrike@webrtc.org
a92fd74f40 (Auto)update libjingle 63773382-> 63775799 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 01:46:18 +00:00
henrike@webrtc.org
dce3feb0b0 (Auto)update libjingle 63738002-> 63773382 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5787 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 01:17:30 +00:00
solenberg@webrtc.org
440fa23553 Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out. 
BUG=2954
R=mflodman@webrtc.org, stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5786 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 19:57:07 +00:00
andrew@webrtc.org
6cd201cf31 Revert 5775 "Modify bitrate controller to update bitrate based o..." 
This triggered an occasional TSAN failure in
CallTest.ReceivesPliAndRecoversWithNack e.g.:
http://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan/builds/1444/steps/memory%20test%3A%20video_engine_tests/logs/stdio

I managed to reproduce this locally and verified that reverting this CL
corrected it.

> Modify bitrate controller to update bitrate based on process call and not
> only whenever a RTCP receiver block is received.
> 
> Additionally:
>  Add condition to only start rampup after a receiver block is received. This was same as old behaviour but now an explicit check is needed to verify process does not ramps up before the first block.
> 
>  Fix logic around capping max bitrate increase at 8% per second. Before it was only increasing once every 1 second and each increase would be as high as 8%. If receiver blocks had a different interval before it would lose an update or waste an update slot and not ramp up as much as a 8% (e.g. if RTCP received < 1 second).
> 
>  Did not touch decrease logic, however since it can be triggered more often it
>  may decrease much faster and closer to the original written cap of once every
>  300ms + rtt.
> 
> Note:
>  rampup_tests.cc don't seem to be affected by this since there is no packet loss or REMB that go higher than expected cap.
>  bitrate_controller_unittests.cc are don't really simulate a clock and the process thread, but trigger update by inserting an rtcp block.
> 
> BUG=3065
> R=stefan@webrtc.org, mflodman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/10529004

TBR=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10079005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5785 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 19:42:39 +00:00
mallinath@webrtc.org
681d448d88 Removing VideoCodecDerived and moving methods inside VideoCodec. 
VideoCodecDerived added to handle changes to talk (fakewebrtcvideoengine.h).

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

Review URL: https://webrtc-codereview.appspot.com/10569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5784 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 18:44:58 +00:00
elham@webrtc.org
39f8ddae70 Updated WebRTC version to 3.51 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5783 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 18:41:14 +00:00
henrike@webrtc.org
ae3347a546 Fix after auto update: removed files were brought back. 
BUG=N/A
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 18:17:02 +00:00
fischman@webrtc.org
e52b3b9c95 iOS video_capture: move @private vars to impl. 
Promised change from https://webrtc-codereview.appspot.com/10539005/ that got
dropped accidentally.

R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5781 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 18:15:07 +00:00
fischman@webrtc.org
76d4f389bb AppRTCDemo(iOS): allow rooms with no incoming audio. 
Also fix a compile-time warning for a leftover unimplemented method
(RTCVideoRenderer:setTransform).

R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5780 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 17:40:38 +00:00
henrike@webrtc.org
6e3dbc2a77 (Auto)update libjingle 63648983-> 63738002 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5779 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 17:09:47 +00:00
sprang@webrtc.org
efcad39f77 Fix race condition in RTPSEnder. 
In RTPSender::SendPayloadType(), payload_type_ should not be read
without owning send_critsect_.

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

Review URL: https://webrtc-codereview.appspot.com/8199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5778 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 16:51:35 +00:00
tina.legrand@webrtc.org
ff7908abfd Roll Opus with ARM optimizations enabled to WebRTC 
This CL roll latest Opus changes from Chromium.

The major update is that optimizations are enabled for ARM processors.

BUG=
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10599005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 16:12:43 +00:00
henrik.lundin@webrtc.org
02e749f848 Change sprintf format string from %zu to %i 
The resulting string became wrong on Windows. Instead of printing
the numerical value in number_of_streams_, the string "zu" got
printed. (Linux and Mac worked fine already.)

This will result in a change of statistics name in the performance
graphs.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10569005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5776 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 13:39:11 +00:00
andresp@webrtc.org
da07737e68 Modify bitrate controller to update bitrate based on process call and not 
only whenever a RTCP receiver block is received.

Additionally:
 Add condition to only start rampup after a receiver block is received. This was same as old behaviour but now an explicit check is needed to verify process does not ramps up before the first block.

 Fix logic around capping max bitrate increase at 8% per second. Before it was only increasing once every 1 second and each increase would be as high as 8%. If receiver blocks had a different interval before it would lose an update or waste an update slot and not ramp up as much as a 8% (e.g. if RTCP received < 1 second).

 Did not touch decrease logic, however since it can be triggered more often it
 may decrease much faster and closer to the original written cap of once every
 300ms + rtt.

Note:
 rampup_tests.cc don't seem to be affected by this since there is no packet loss or REMB that go higher than expected cap.
 bitrate_controller_unittests.cc are don't really simulate a clock and the process thread, but trigger update by inserting an rtcp block.

BUG=3065
R=stefan@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5775 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 12:48:42 +00:00
sprang@webrtc.org
0f0c992336 Temporarily use older protobuf library. 
BUG=3106
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5774 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 12:43:58 +00:00
stefan@webrtc.org
a16147c037 Adding API for setting bandwidth estimation configurations. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5773 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 10:37:31 +00:00
fischman@webrtc.org
b64d52c292 iOS video_capture: start camera in the background. 
Camera start is a blocking operation so never a good idea to do on a main
thread, but worse than that is that the guts of WebView appear to be
interacting with capture start in a bad way causing startup to pause for 10s
while a timeout expires.  This change eliminates that 10s delay.

R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5772 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 05:23:32 +00:00
fischman@webrtc.org
385a722646 PeerConnection(iOS): make ARC-clean talk/.../objc* and talk/examples/ios 
- Removes a strong-reference cycle between RTCPeerConnection and
  RTCPeerConnectionObserver
- Gives RTCPeerConnectionObserver a virtual dtor
- Ensures RTCPeerConnectionTest tears down correctly
- Ensures AppRTCDemo tears down correctly

This is the talk/ half; the webrtc/ half is in https://webrtc-codereview.appspot.com/10539005

BUG=3054,3055,3100
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10499005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5771 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 05:16:29 +00:00
fischman@webrtc.org
e68102e046 iOS VideoEngine: move video_{capture,render} to ARC. 
Replaces ye olde timey explicit release with teh hotness of automatic
reference counting.

This is the webrtc/ half; the talk/ half is in https://webrtc-codereview.appspot.com/10499005/

BUG=3054,3055
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10539005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5770 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 05:15:44 +00:00
sergeyu@chromium.org
e42b8ab129 Cleanups in libjingle to make it compile with chromium_code=1 
Fixed all warnings that show up when compiling libjingle
in chromium with compiling with chromium_code=1.
chromium_code=1 enables various warnings that are off by
default. Most changes are for unused variables and consts.

R=pthatcher@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5769 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 00:31:35 +00:00
fischman@webrtc.org
7fa1fcb72c AppRTCDemo(ios): style/cleanup fixes following cr/62871616-p10 
BUG=2168
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/9709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 00:11:56 +00:00
asapersson@webrtc.org
ce12f1fd32 Add configuration for ability to use the encode usage measure for triggering overuse/underuse. 
BUG=1577
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5767 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-24 21:59:16 +00:00
andrew@webrtc.org
b70c8e9dfd Disable flaky WebRtcVideoMediaChannelTests on memcheck and tsan. 
BUG=3096
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5766 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-24 20:57:42 +00:00
solenberg@webrtc.org
3fb8f7bbb0 Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5765 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-24 20:28:11 +00:00
fischman@webrtc.org
c693a2a624 PeerConnection(iOS): fix case in #import statements. 
We've been skating by on OS/X's default case-insensitive filesystem, but this
is a bit silly.

This change brought to you by:
sed -i '' 's/\+internal\.h/+Internal.h/g' $(git grep -l '+internal.h')

BUG=3088
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5764 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-24 18:56:37 +00:00
stefan@webrtc.org
9d4762e8b6 Have changes to REMB trigger RTCP to be sent immediately. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5763 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-24 17:13:00 +00:00