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Author SHA1 Message Date
hclam@chromium.org
806dc3b0e6 More trace events 
The goal of this change is to unify tracing events styles
and add trace events for all RTP traffic.

BUG=1555
Review URL: https://webrtc-codereview.appspot.com/1290007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 19:54:10 +00:00
stefan@webrtc.org
4d2f5de67a Improve how NACK lists are generated before a frame has been decoded. 
BUG=1598

Review URL: https://webrtc-codereview.appspot.com/1295004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 18:24:41 +00:00
pbos@webrtc.org
ac891627c6 WebRtc_Word32 -> int32_t in audio_conference_mixer/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1306004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3804 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 17:40:15 +00:00
pbos@webrtc.org
b09130763b WebRtc_Word32 -> int32_t in common_audio/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3803 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 16:40:28 +00:00
stefan@webrtc.org
7da3459b2a Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps." 
This reverts commit 4954b3650192d78037714138a5c519ef08f2670e.
Reverts r3799

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1308004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 14:56:29 +00:00
pbos@webrtc.org
b238d1210b WebRtc_Word32 -> int32_t in video_engine/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1302005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3801 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:41:51 +00:00
pbos@webrtc.org
1ab45f6dd5 WebRtc_Word32 -> int32_t in video_processing/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1297006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3800 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:38:10 +00:00
stefan@webrtc.org
afcc6101d0 With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps. 
We should consider making the same change to the render timestamps generated at the receiver.

BUG=1563

Review URL: https://webrtc-codereview.appspot.com/1283005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:37:40 +00:00
pbos@webrtc.org
fd2bfc8fca WebRtc_Word32 -> int32_t in common_video. 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1300004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3798 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:36:08 +00:00
pbos@webrtc.org
c75102eba7 WebRtc_Word32 -> int32_t in utility/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1307005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3797 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:32:55 +00:00
pbos@webrtc.org
0ea11c1768 WebRtc_Word32 -> int32_t in media_file/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1304005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3796 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:31:37 +00:00
hta@webrtc.org
a701c0ed03 Fixing the flakiness of ThreadWakesTwice. 
TESTED=ran the test 10.000 times with machine load.
BUG=1270

Review URL: https://webrtc-codereview.appspot.com/1303004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3795 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 12:36:28 +00:00
pbos@webrtc.org
a5f1787f63 WebRtc_Word32 -> int32_t in test/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1302004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3794 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 11:10:21 +00:00
pbos@webrtc.org
2550988baa WebRtc_Word32 -> int32_t in audio_device/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1302006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3793 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 10:30:35 +00:00
pbos@webrtc.org
6141e13873 WebRtc_Word32 -> int32_t in voice_engine/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1305004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3792 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 10:09:10 +00:00
pbos@webrtc.org
046deb9b20 WebRtc_Word32 -> int32_t in system_wrappers 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1301004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3791 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 09:06:11 +00:00
pbos@webrtc.org
29758de9b6 Always set render delay in ViEChannel::RegisterExternalDecoder. 
BUG=1523

Review URL: https://webrtc-codereview.appspot.com/1219007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3790 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 00:34:42 +00:00
pbos@webrtc.org
0946a56023 WebRtc_Word32 => int32_t etc. in audio_coding/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1271006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 00:28:06 +00:00
pwestin@webrtc.org
6faf71d27b Remove the old unused udp_transport 
Review URL: https://webrtc-codereview.appspot.com/1272009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 23:25:25 +00:00
vikasmarwaha@webrtc.org
4c44fe0561 Updated pranswer, dtmf demos & deleted pc1-deprecated.html. 
Review URL: https://webrtc-codereview.appspot.com/1287007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3783 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 21:23:58 +00:00
marpan@webrtc.org
6ff76c7404 Reduce execution time of rate control test. 
TBR=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1289005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 20:32:48 +00:00
kma@webrtc.org
cf8e108158 Fixed a bug in isac-fix's entropy coding function: out of bounds acces to array. 
BUG=227286
Review URL: https://webrtc-codereview.appspot.com/1293005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3781 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 16:37:53 +00:00
pbos@webrtc.org
b4a0623e43 Fix of lint script errors in apprtc.py 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1285007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3780 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 15:59:24 +00:00
pbos@webrtc.org
f2e7bc6b6a Added maxlen=80 to CheckLongLines() call in PRESUBMIT.py 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1285006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3779 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 15:46:07 +00:00
pbos@webrtc.org
034f004a4f WebRtc_Word32 => int32_t in video_coding/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1203008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3778 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 11:13:29 +00:00
pbos@webrtc.org
2f44673d66 WebRtc_Word32 => int32_t for rtp_rtcp/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1279007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 11:08:41 +00:00
mflodman@webrtc.org
367804cce2 Clean packets on the network when closing + made loopback test actually run again. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1290006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3776 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 10:42:50 +00:00
pbos@webrtc.org
ff7e1303e8 WebRtc_Word32 => int32_t remote_bitrate_estimator/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1275009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3775 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 10:04:37 +00:00
hta@webrtc.org
37bf5847dc Show stats from both sides 
This change shows the stats generated both at the sending PeerConnection
and at the receiving PeerConnection.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1290005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3774 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-06 10:05:55 +00:00
vikasmarwaha@webrtc.org
222e9948f5 Migrating Apprtc to use new TURN service which supports time-limited TURN credentials. 
Review URL: https://webrtc-codereview.appspot.com/1291004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3773 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-06 05:58:15 +00:00
wu@webrtc.org
123b618f48 Fix a crash issue on WinXP where LoadLibrary(TEXT("Kernel32.dll")) may fail. 
BUG=crbug.com/226301
TBR=henrike
Review URL: https://webrtc-codereview.appspot.com/1293004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3772 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-06 04:13:51 +00:00
turaj@webrtc.org
2e6b7e938f In streaming mode it is preferable to fade to silence when sender stops sending, or long period of packet loss. 
test=try bots.
Review URL: https://webrtc-codereview.appspot.com/1272004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3771 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-06 00:08:11 +00:00
henrika@webrtc.org
19da719a5f Resolves TSan v2 reports data races in voe_auto_test. 
--- Note that I will add more fixes to this CL ---

BUG=1590

Review URL: https://webrtc-codereview.appspot.com/1286005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3770 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-05 14:34:57 +00:00
kjellander@webrtc.org
10eb92039b Add GYP target for WebRTC Video demo for Android. 
Add a build target for the Video demo app for Android that only
exists when OS=='android' during build.

Note that this doesn't solve webrtc:1029, it's more like a workaround
waiting for the complete solution, which is to great a proper GYP target
that doesn't involve an action and an external script.

BUG=1029
TEST=Built successfully with:
source build/android/envsetup.sh
gclient runhooks
ninja -C out/Debug
Also verified the target is not present when OS is not 'android'.

Review URL: https://webrtc-codereview.appspot.com/1286004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3769 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-05 13:36:32 +00:00
pbos@webrtc.org
b5bf54c4e7 Permit arbitrary payload names for kVideoCodecGeneric. 
BUG=1575

Review URL: https://webrtc-codereview.appspot.com/1282005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-05 13:27:38 +00:00
pwestin@webrtc.org
b9e402d99f Remove WEBRTC_*_ENGINE_NETWORK_API use 
Review URL: https://webrtc-codereview.appspot.com/1203009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3767 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 19:51:42 +00:00
edjee@google.com
79b0289bfc Adds event traces and counters for WebRTC receive side. 
Review URL: https://webrtc-codereview.appspot.com/1279005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3766 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 19:43:34 +00:00
pwestin@webrtc.org
835dbf4516 Fix no received audio in tests. 
BUG=1582, 1581
Review URL: https://webrtc-codereview.appspot.com/1281005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3763 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 17:24:15 +00:00
henrika@webrtc.org
aa527bbc91 Disabling MixingTests due to race conditions. 
BUG=1580
TBR=tommi

Review URL: https://webrtc-codereview.appspot.com/1285005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3762 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 15:19:10 +00:00
hta@webrtc.org
fcb7c38b15 Two more sleep calls converted to use SleepMs(). 
This is CL 753005 in its new home.

BUG=603

Review URL: https://webrtc-codereview.appspot.com/1201008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3761 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 08:56:34 +00:00
henrika@webrtc.org
bb8ada686e TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC 
BUG=226044
TEST=content_unittests in Chrome with TSan v2 enabled

Review URL: https://webrtc-codereview.appspot.com/1201010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3760 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 08:39:09 +00:00
pwestin@webrtc.org
0c45957e3a Remove UDP transport API from VoE 
Review URL: https://webrtc-codereview.appspot.com/1236004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3757 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-03 15:43:57 +00:00
henrika@webrtc.org
0746ce1465 Fixes memory leak in AudioLevel class reported by memory try bots. 
TBR=tommi

Review URL: https://webrtc-codereview.appspot.com/1275008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3756 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-03 11:58:12 +00:00
henrika@webrtc.org
d108a46206 Fixes data race in WebRTCAudioDeviceTest.StartRecording reported by ThreadSanitizer 
BUG=225690

Review URL: https://webrtc-codereview.appspot.com/1269008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3755 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-03 11:25:31 +00:00
pwestin@webrtc.org
82dcc9ff11 Remove UDP transport API from ViE 
Review URL: https://webrtc-codereview.appspot.com/1232004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3754 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 20:37:14 +00:00
pbos@webrtc.org
7b859cc1e9 Webrtc_Word32 => int32_t in video_coding/main/ 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3753 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 15:54:38 +00:00
henrike@webrtc.org
cfc07c943f Revert of r3747. 
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1277005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3752 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 14:55:44 +00:00
hta@webrtc.org
95d88735ee Two more sleep calls converted to use SleepMs(). 
BUG=603

Review URL: https://webrtc-codereview.appspot.com/753005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3751 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 14:46:33 +00:00
henrika@webrtc.org
4ff956f428 Fixes data race in WebRTCAudioDeviceTest.Construct reported by ThreadSanitizer 
BUG=159112

Review URL: https://webrtc-codereview.appspot.com/1201007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3750 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 11:59:11 +00:00
kjellander@webrtc.org
46e626d3b8 Fix gflags compile error on x86 Android 
This CL is the landing of http://review.webrtc.org/1277004/ for yujie.mao@intel.com.

I verified the added files are identical with the previously added ones
in third_party/google-gflags/gen/arch/linux/ia32 (which is the way this library needs to be handled when supporting the additional Android platforms).

BUG=none
TEST=Successfully compiled WebRTC on Linux Precise with:
source build/android/envsetup.sh --target-arch=x86
gclient runhooks
ninja -C out/Debug

Review URL: https://webrtc-codereview.appspot.com/1273005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3749 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 11:07:04 +00:00