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885cd13356
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Start NACKing as soon as we have the first packet of a key frame.
BUG=1605
Review URL: https://webrtc-codereview.appspot.com/1307007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3855 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-16 09:38:26 +00:00 |
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bdb9b971be
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Change receive statistics bitrate to be provided in bps instead of kbps.
BUG=1469
Review URL: https://webrtc-codereview.appspot.com/1326004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3854 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-16 09:02:03 +00:00 |
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92d1f07551
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Elevate NetEq short-term activity statistics to ACM level for logging.
Review URL: https://webrtc-codereview.appspot.com/1313004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3850 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-15 16:52:04 +00:00 |
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4b8de90dce
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Disable -Wunsequenced warning in audio_coding_module
BUG=1647
TEST=Compile locally on Linux with clang enabled.
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1316005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3848 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-15 06:38:56 +00:00 |
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c2a3aa7926
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Partial revert of r3844
Review URL: https://webrtc-codereview.appspot.com/1320004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3845 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-12 19:53:30 +00:00 |
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d6bd7cd2b1
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removing redundant calls to cleanframes
Review URL: https://webrtc-codereview.appspot.com/1318004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3844 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-12 17:09:51 +00:00 |
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9f5ebb5251
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Adding a payload type for RTX.
BUG=736
TEST=Modified RTP unittests.
Review URL: https://webrtc-codereview.appspot.com/1278004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3843 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-12 14:55:46 +00:00 |
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b8e7f4cc97
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Change capture interface to use NTP capture time.
Move NTP functionality to Clock.
BUG=1563
TEST=trybots and vie_auto_test --automated
Review URL: https://webrtc-codereview.appspot.com/1313005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3842 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-12 11:56:23 +00:00 |
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9da751715f
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VCM/JB:Removing hybrid and setting a decodable state.
Review URL: https://webrtc-codereview.appspot.com/1283004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3834 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-11 18:49:13 +00:00 |
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7bc465bd21
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Fix issues with incorrect wrap checks when having big buffers and high bitrate.
Introduces shared functions for timestamp and sequence number wrap checks.
BUG=1607
TESTS=trybots
Review URL: https://webrtc-codereview.appspot.com/1291005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3833 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-11 17:48:02 +00:00 |
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122d209e67
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Fixes an issue where the start bitrate is stored in kbps instead of bps.
BUG=1638
TEST=trybots and vie_auto_test loopback with nack.
Review URL: https://webrtc-codereview.appspot.com/1312004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3831 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-11 17:21:40 +00:00 |
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eac36b8561
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Fix -Wstring-conversion warnings.
Review URL: https://webrtc-codereview.appspot.com/1299007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3830 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-11 15:37:46 +00:00 |
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523f93729b
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Re-write the build of the nacklist.
Review URL: https://webrtc-codereview.appspot.com/1304008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3822 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-11 11:30:39 +00:00 |
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f2a97fc2b4
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WebRTCDemo: handle stride!=width from first frame.
Previously only mid-stream frames handled stride!=width correctly.
BUG=1615
Review URL: https://webrtc-codereview.appspot.com/1304009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3821 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-10 23:21:10 +00:00 |
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e4b6064f8e
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Replace legacy G_CONST with const.
BUG=1608
Review URL: https://webrtc-codereview.appspot.com/1310005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3814 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-10 18:06:57 +00:00 |
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ab9202b673
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Removing remaining WebRtc_Word32 not in typedefs.h
BUG=
Review URL: https://webrtc-codereview.appspot.com/1306006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3813 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-10 17:59:17 +00:00 |
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77d59fe408
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WebRTCDemo: no-op out instead of NPEing on destroyed camera.
BUG=1617
Review URL: https://webrtc-codereview.appspot.com/1310004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3812 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-10 17:11:51 +00:00 |
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dfc5bb9c97
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WebRtc_Word32 -> int32_t in video_capture/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1298005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3811 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-10 08:23:13 +00:00 |
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ddf94e71e5
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WebRtc_Word32 -> int32_t in video_render/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1304006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3810 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-10 08:09:04 +00:00 |
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b7192b8247
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WebRtc_Word32 -> int32_t in audio_processing/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1307004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3809 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-10 07:50:54 +00:00 |
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557e92515d
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Reapply the reverted r3747.
https://code.google.com/p/webrtc/source/detail?r=3747
r3747 timed-out on a tsan test. Verified that it passes
the test and reduced the execution time of that test (r3782).
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1292006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3807 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-09 21:21:32 +00:00 |
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806dc3b0e6
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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
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2013-04-09 19:54:10 +00:00 |
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4d2f5de67a
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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
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2013-04-09 18:24:41 +00:00 |
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ac891627c6
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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
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2013-04-09 17:40:15 +00:00 |
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7da3459b2a
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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
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2013-04-09 14:56:29 +00:00 |
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1ab45f6dd5
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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
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2013-04-09 13:38:10 +00:00 |
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afcc6101d0
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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
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2013-04-09 13:37:40 +00:00 |
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c75102eba7
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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
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2013-04-09 13:32:55 +00:00 |
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0ea11c1768
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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
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2013-04-09 13:31:37 +00:00 |
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2550988baa
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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
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2013-04-09 10:30:35 +00:00 |
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0946a56023
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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
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2013-04-09 00:28:06 +00:00 |
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6faf71d27b
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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
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2013-04-08 23:25:25 +00:00 |
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6ff76c7404
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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
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2013-04-08 20:32:48 +00:00 |
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cf8e108158
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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
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2013-04-08 16:37:53 +00:00 |
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034f004a4f
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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
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2013-04-08 11:13:29 +00:00 |
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2f44673d66
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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
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2013-04-08 11:08:41 +00:00 |
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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
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2013-04-08 10:04:37 +00:00 |
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2e6b7e938f
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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
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2013-04-06 00:08:11 +00:00 |
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19da719a5f
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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
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2013-04-05 14:34:57 +00:00 |
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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
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2013-04-05 13:27:38 +00:00 |
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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
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2013-04-04 19:43:34 +00:00 |
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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
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2013-04-04 08:39:09 +00:00 |
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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 |
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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
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2013-04-02 14:55:44 +00:00 |
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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
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2013-04-02 14:46:33 +00:00 |
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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
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2013-04-02 11:59:11 +00:00 |
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f81fad6267
|
Fix opus bitrate truncated to 16-bit int. This prevented setting bitrates higher
than 2^16kbps.
Review URL: https://webrtc-codereview.appspot.com/1275004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3748 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-01 22:25:11 +00:00 |
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747c4cc96e
|
For VGA (640x360), currently 1 thread is used. This change increases it to 2 threads. For HD, 4 threads are enabled.
BUG=none
TEST=run a hangout and screencast high framerate, high resolution windows of youtube. Observe that 1 cpu is insufficient to maintain high framerate with complex content.
Review URL: https://webrtc-codereview.appspot.com/1203006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3747 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-01 22:16:45 +00:00 |
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29f34b8727
|
Fix for issue: https://code.google.com/p/webrtc/issues/detail?id=1549
Review URL: https://webrtc-codereview.appspot.com/1270004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3741 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-03-28 18:57:46 +00:00 |
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93bea51517
|
Removed CPU APIs from VoEHardware. Code is now only used by test applications.
Recommitting https://code.google.com/p/webrtc/source/detail?r=3736 after fixing build break.
BUG=8404677
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3739 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-03-28 15:58:49 +00:00 |
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