388bd79a76Update checksums for AcmReceiverBitExactness on android
henrik.lundin@webrtc.org
2014-08-13 10:38:15 +00:00
023f12fb6eNetEq background noise generation off by default
henrik.lundin@webrtc.org
2014-08-13 09:45:40 +00:00
c27543d297Fix STAP-A bug where we might overflow the packet buffer due to not accounting for the length of the length field.
stefan@webrtc.org
2014-08-13 07:40:45 +00:00
e999bd087bRemoving ASSERT for tcp candidate for port 0 and 9, as Android clients may not be called with set_allow_tcp_listen(false).
mallinath@webrtc.org
2014-08-13 06:05:55 +00:00
afb554f404Move default-recv-channels to a separate class.
pbos@webrtc.org
2014-08-12 23:17:13 +00:00
c891fee7abMake a int64 constant use ULL suffix so it wont get truncated. BUG=3690 TESTED=try bots R=andrew@webrtc.org
fbarchard@google.com
2014-08-12 22:39:06 +00:00
c6273b53ddDrMemory suppresssions, likely from r6811.
marpan@webrtc.org
2014-08-12 21:29:06 +00:00
cf8f33a6d6Removes mismatching signs in signal_processing_unittests
bjornv@webrtc.org
2014-08-12 10:27:21 +00:00
6aac93bd9cAdding SetOpusMaxBandwidth in VoE and ACM
minyue@webrtc.org
2014-08-12 08:13:33 +00:00
c98ce3b34cmodules/audio_processing: Updates output_data_fixed.pb test file
bjornv@webrtc.org
2014-08-12 07:35:52 +00:00
6ac22e6b47Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798
henrike@webrtc.org
2014-08-11 21:06:30 +00:00
820f8e9ca7modules/audio_processing: Moves declaration of kDelayDiffOffsetSamples
bjornv@webrtc.org
2014-08-11 15:39:00 +00:00
4e4b0984daMerge NetEqDecodingTest.TestBitExactnesst and .TestNetworkStatistics
henrik.lundin@webrtc.org
2014-08-11 14:48:49 +00:00
065247b5b7Rebase webrtc/base with r6863 version of talk/base: cls integrated: r6809 svn diff -r 6808:6809 http://webrtc.googlecode.com/svn/trunk/talk/base > 6809.diff patch -p0 -i 6809.diff
henrike@webrtc.org
2014-08-11 14:32:13 +00:00
730bf30da7Refactor 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.
tommi@webrtc.org
2014-08-11 14:08:33 +00:00
1c8391205eUse test::Packet test::PacketSource classes in neteq_rtpplay
henrik.lundin@webrtc.org
2014-08-11 12:29:38 +00:00
96d8b0e69fRevert 6860 "SSE2 version of SubbandCoherence()"
bjornv@webrtc.org
2014-08-11 12:09:13 +00:00
0db82f337fSSE2 version of SubbandCoherence()
bjornv@webrtc.org
2014-08-11 10:38:31 +00:00
7ec3f9f838Fix a bug in parsing IceCandidate with IPV6 address. It used to treat ":" as a candidate delimiter and got confused by the ":" in the IPV6 address. The new logic is to check if the input has multiple lines. If so, returns error.
jiayl@webrtc.org
2014-08-08 23:09:15 +00:00
59a2f9f584Remove the old H264 code now that a new H.264 packetizer has been implemented.
stefan@webrtc.org
2014-08-07 15:09:24 +00:00
9d74f7ce8cFix single nalu packetization bug.
stefan@webrtc.org
2014-08-07 15:02:16 +00:00
e8c84bf4deFix so video_replay logs aren't spammed.
pbos@webrtc.org
2014-08-07 14:42:45 +00:00
1d956dd1a7Since the packet loss rate cannot be estimated accurately, there is always a mismatch between the estimated packet loss rate and the true one. Such a mismatch will make Opus FEC suboptimal.
minyue@webrtc.org
2014-08-07 12:31:36 +00:00
ea25784107Change how background noise mode in NetEq is set
henrik.lundin@webrtc.org
2014-08-07 12:27:37 +00:00
e415864a32GN: Add PRESUBMIT.py check for GN changes + default bots.
kjellander@webrtc.org
2014-08-06 09:11:18 +00:00
8b033adb19Change the way we reference enumerators in H.264 packetization code to be standard C++ compliant.
stefan@webrtc.org
2014-08-06 08:06:53 +00:00
56d8e05238A followup to r6828 to fix a condition check in mediasession.cc.
jiayl@webrtc.org
2014-08-05 23:52:36 +00:00
d7b4dea801initialize packet len in NETEQTEST_DummyRTPpacket.cc and NETEQTEST_RTPpacket.cc to fix build error on vs2013 BUG=3660 TESTED=set DEPOT_TOOLS_WIN_TOOLCHAIN=0 & set GYP_DEFINES=target_arch=ia32 & call python webrtc\build\gyp_webrtc -G msvs_version=2013 &ninja -C out\Debug R=andrew@webrtc.org
fbarchard@google.com
2014-08-05 23:46:42 +00:00
dde16f19e3Fix some code styles.
pbos@webrtc.org
2014-08-05 23:35:43 +00:00
e7d47a1473Maintain the order of the m-lines in CreateOffer and CreateAnswer. The order in the offer follows the order in the current local description. The order in the answer follows the order in the current offer.
jiayl@webrtc.org
2014-08-05 19:19:05 +00:00
e086af0fa3Fix implicite cast from signed int to unsigned int in unittest.cc BUG=3636 TESTED=set GYP_DEFINES=target_arch=ia32 & call python webrtc\build\gyp_webrtc -G msvs_version=2013 & ninja -C out\Debug R=pthatcher@webrtc.org
fbarchard@google.com
2014-08-05 17:10:52 +00:00
d6542852f3Unbreaks linux.cc in Chromium.
henrike@webrtc.org
2014-08-04 21:51:14 +00:00
b18bf5e47dAdds the support of RTCOfferOptions for PeerConnectionInterface::CreateOffer. Constraints are still supported for CreateOffer, but converted to RTCOfferOptions internally.
jiayl@webrtc.org
2014-08-04 18:34:16 +00:00
b01ce14b13add some comments about DEPS lkgr for chromium BUG=none TESTED=none R=harryjin@google.com
fbarchard@google.com
2014-08-04 18:07:19 +00:00
c9b507253fDrMemory suppression due to r6811.
henrike@webrtc.org
2014-08-04 16:48:24 +00:00
ee135f78b7Memcheck suppression. Re-suppress 3478 suppression after namespace change from talk_base to rtc.
henrike@webrtc.org
2014-08-04 15:35:14 +00:00
84b9e1e9d9Fix for retransmission. Base layer packets were not retransmitted. Issue introduced in r6669.
asapersson@webrtc.org
2014-08-04 11:59:42 +00:00
42d65ce8d7Fix memory leak in FakeSSLCertificate::GetChain(), discovered by Linux Memcheck build/try bots.
solenberg@webrtc.org
2014-08-01 10:07:46 +00:00
bfe6e08195Add simulation of network effects to video_loopback tool.
stefan@webrtc.org
2014-07-31 12:30:18 +00:00
d9843da9eelibjingle: stop building files in talk/base as they are no longer used as of r6799
henrike@webrtc.org
2014-07-30 04:00:52 +00:00
48305f5f4cDisable warning 4702 which affects map, xlist and others on vs2012 and vs2013. BUG=3584 TESTED=python webrtc\build\gyp_webrtc -G msvs_version=2013 & ninja -C out\Release R=pthatcher@webrtc.org
fbarchard@google.com
2014-07-30 00:16:20 +00:00
901debdad3roll libyuv to r1038 from r1035 to add gyp define that makes jpeg optional. BUG=libyuv:346 TESTED=set GYP_DEFINES=target_arch=ia32 libyuv_disable_jpeg=1 R=andrew@webrtc.org
fbarchard@google.com
2014-07-29 18:07:07 +00:00
fc8b0871d9Remove dependency on openssl for android, add dependency on boringssl. Should make Android bots green again.
solenberg@webrtc.org
2014-07-29 15:23:59 +00:00
fdbe1442c5Use C functions in aec for MIPS
andrew@webrtc.org
2014-07-29 14:39:10 +00:00
e75d78d32dIntegrate rtcp packet class to rtcp receiver tests.
asapersson@webrtc.org
2014-07-29 08:21:50 +00:00
1e7d60e451merge_libs.py: fixes Windows breakage: there should be no space after "lib /OUT:".
henrike@webrtc.org
2014-07-29 04:45:23 +00:00
961293d469webrtc/base: FileModifyTime -> OlderThan as that's what it was ever used as. Needed for cl/70828325.
henrike@webrtc.org
2014-07-25 21:58:50 +00:00
af9e7943d1Fix compilation on windows with clang, indentation cleanups
sergeyu@chromium.org
2014-07-25 19:42:19 +00:00
257e130a16Set NACK/REMB when setting receive codecs.
pbos@webrtc.org
2014-07-25 19:01:32 +00:00
185636cf70Revert of 6778 "Refactor StatsCollector and associated types." Breakes FYI bots.
henrike@webrtc.org
2014-07-25 18:44:42 +00:00
c7b8f39e2eFixes "argument list too long" problem on Linux by using the "find" command instead of re-implementing one in python.
henrike@webrtc.org
2014-07-25 18:36:55 +00:00