43a1395370AppRTCDemo(android): README updates for a shrinking envsetup.sh world.
fischman@webrtc.org
2014-05-28 17:29:09 +00:00
b364016cbbRevert r6161 "Drop the DataChannel message if it's received when the channel is not open." The spec does not say the DataChannel has to be open to receive a message.
jiayl@webrtc.org
2014-05-28 16:37:25 +00:00
f15c14be22Echo canceler: Saturate output to guarantee it'll be in the allowed range
kwiberg@webrtc.org
2014-05-28 11:47:08 +00:00
c1a40a7b68This CL is to adding feedback of packet loss rate to encoder in voice engine. A direct reason for doing it is to make use of Opus FEC, which can adapt itself to changes in the packet loss rate.
minyue@webrtc.org
2014-05-28 09:52:06 +00:00
aca5939dfccommon_audio/signal_processing: Fixes arm compilation issues with gcc 4.8
bjornv@webrtc.org
2014-05-28 08:45:04 +00:00
0aa3ee661cBetter buffer size estimation in NetEq for redundant packets
minyue@webrtc.org
2014-05-28 07:48:01 +00:00
ab6bf4f54cAdded api for getting cpu measures using a struct.
asapersson@webrtc.org
2014-05-27 07:43:15 +00:00
74767401f2Fix a bug preventing FilePlayer from playing encoded wav files
henrik.lundin@webrtc.org
2014-05-26 13:37:45 +00:00
1457b4737aFirst incoming packet was not accounted for in receive stats. Changed call order for incoming packet to receive statistics class. Receive stats is reset if the payload type changes. Update stats after a possible reset.
asapersson@webrtc.org
2014-05-26 13:06:04 +00:00
39eccefbdeDisable ChannelManagerTest.StartupShutdownOnUnstartedThread The test is testing a scenario that shouldn't happen.
fischman@webrtc.org
2014-05-23 17:50:38 +00:00
e5063b1733Thread: delete racy API (Release()) and fix racy code (started()).
fischman@webrtc.org
2014-05-23 17:28:50 +00:00
18f41b8eb4PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
fischman@webrtc.org
2014-05-23 17:27:18 +00:00
546961a9d3Avoid reading uninitialized values (outside baundary) in DFT arithmatic decoder of iSAC-fix.
turaj@webrtc.org
2014-05-23 17:14:29 +00:00
aa5ea1c0f91. Make a clear distinction between codec internal FEC and RED, confusing mentioning of FEC in the old codes is replaced by RED
minyue@webrtc.org
2014-05-23 15:16:51 +00:00
706152dcc9Fix uninitialized reads in IsDefaultBrowserFirefox
pbos@webrtc.org
2014-05-23 14:35:48 +00:00
10f871f29bRemove the restriction to allow having both webrtc and talk changes in the same cl.
wu@webrtc.org
2014-05-22 22:35:46 +00:00
0720758f9fBump WebRTC version number to 3.54 TBR=wu@webrtc.org
tnakamura@webrtc.org
2014-05-22 17:51:18 +00:00
1bb5da04feAdds missing include of assert header.
henrike@webrtc.org
2014-05-22 14:31:14 +00:00
21f7d6d2feWebRTCDemo: move the deletion of CritSect to end of the dtor to fix a crash in Android video renderer.
braveyao@webrtc.org
2014-05-22 02:57:55 +00:00
8e755c1ad2Connect SignalDestroyed in AllocationSequence after TURN ports are destroyed when TURN ports are using shared socket with UDP port.
mallinath@webrtc.org
2014-05-21 23:00:46 +00:00
88fbb2d86bSwitch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
henrike@webrtc.org
2014-05-21 21:18:46 +00:00
a148704b4bRename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h.
henrike@webrtc.org
2014-05-21 16:52:14 +00:00
7ca277b574Initializes WINDOWPLACEMENT::length in GetCroppedWindowRect.
jiayl@webrtc.org
2014-05-21 16:02:31 +00:00
000658a138Revert of 6211 as it was committed despite of PRESUBMIT.py warning. The commit breaks the sync bot.
henrike@webrtc.org
2014-05-21 16:01:13 +00:00
1a79bb8d30WebRTCDemo: clean the error message due to API clean up and add ability to route the audio through all three outputs, headset/earpiece/loudspeaker
braveyao@webrtc.org
2014-05-21 03:37:45 +00:00
125ffd709dSwitch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
henrike@webrtc.org
2014-05-20 15:20:44 +00:00
4059c2f579Disable CallPerfTest.CaptureNtpTimeWithNetworkDelay due to being flaky.
stefan@webrtc.org
2014-05-20 15:12:29 +00:00
70bb2d5755Revert r6198 "Expose the original packet length in in the RTP play tools."
stefan@webrtc.org
2014-05-20 13:25:48 +00:00
83599cba77Reenable WebRtcVideoEngineTestFake.SendReceiveBitratesStats under DrMemory.
stefan@webrtc.org
2014-05-20 13:16:35 +00:00
e208458643Expose the original packet length in in the RTP play tools.
stefan@webrtc.org
2014-05-20 13:09:16 +00:00
be4ab99a53Disabling RealFFTTest.RealAndComplexMatch and AudioProcessingTest.Formats as they currently are broken with gcc 4.8.
stefan@webrtc.org
2014-05-20 12:42:01 +00:00
a36db970bdSuppress GMOCK printouts from TestVideoSenderWithVp8
henrik.lundin@webrtc.org
2014-05-20 11:16:10 +00:00
f3e1341da7VoEVolumeTest: Enabled Linux flaky tests
bjornv@webrtc.org
2014-05-20 10:43:42 +00:00
a826006132Add NACK and RPSI packet types to RTCP packet builder. Fixes bug found when parsing received RPSI packet.
asapersson@webrtc.org
2014-05-20 09:53:51 +00:00
2db9f45038Reduce flakiness of voe_auto_test MixingTest by checking dumped audio size
minyue@webrtc.org
2014-05-20 08:33:30 +00:00
1732a591e7Add a UIView for rendering a video track.
tkchin@webrtc.org
2014-05-19 23:26:01 +00:00
7ca1edb31dRemove IOKit linkage from iOS builds. IOKit has been removed in iOS7, so link fails. iOS build succeeds after removing this setting and the corresponding one in talk/libjingle.gyp. Presubmit script tells me that CLs aren't allowed to touch both talk/ and webrtc/ at the same time so doing this separately.
tkchin@webrtc.org
2014-05-19 21:05:10 +00:00
40bc7779aatalk_base: remove lock inversion between MessageQueue and MessageQueueManager.
fischman@webrtc.org
2014-05-19 17:58:04 +00:00
cb711f77d2Add interface to propagate audio capture timestamp to the renderer.
wu@webrtc.org
2014-05-19 17:39:11 +00:00
64339a7069Don't crash if a frame returned from the decoder is too old.
stefan@webrtc.org
2014-05-19 13:31:35 +00:00
725e582461Use the new gyp_var_prefix local variable set by gyp instead of the global GYP_VAR_PREFIX set by the makefiles, since the latter is not guaranteed to still be the same value at the time the command is executed. Also, use abspath instead of realpath to convert paths to absolute, since realpath expands to the empty string if the target file doesn't exist, complicating build debugging.
michaelbai@google.com
2014-05-16 17:56:10 +00:00
14abcc7322libvpx's UNUSED macro conflicts with webrtc/base's. Added missing include of assert.h. Globally defined function "Unused" in talk/base and its copy (webrtc/base) is causing a conflict.
henrike@webrtc.org
2014-05-16 16:54:44 +00:00
1e019d10b8Fix delivery error-checking missed in r6151.
pbos@webrtc.org
2014-05-16 11:38:45 +00:00
57e060251aFix flaky test SendRtpRtcpHeaderExtensionsTest.SentPackets*.
solenberg@webrtc.org
2014-05-16 11:27:09 +00:00
60015d27aeWire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main}
andresp@webrtc.org
2014-05-16 09:39:51 +00:00
75718cf80a* Implement WindowsRealTimeClock::CurrentTimeVal with GetSystemTimeAsFileTime as it supposes to return a POSIX gettimeofday, so that later it can be converted to NTP timee correctly. * Updated to ClockTest.NtpTime to verify the returned NTP is at least larger than kNtpJan1970.
wu@webrtc.org
2014-05-15 23:54:14 +00:00
bf58a75dd9removed webrtc_base_tests_utils from merge libs as it was breaking some builds.
henrike@webrtc.org
2014-05-15 21:45:09 +00:00
508795f088Made the presubmit script accept license headers back to 2003
henrike@webrtc.org
2014-05-15 18:21:17 +00:00
b4e80e095fRe-enable almost all NetEqDecodingTests for Android
henrik.lundin@webrtc.org
2014-05-15 07:14:00 +00:00
7cb4752184WebRTCDemo: couldn't run a second time. The reason is voe could register/unregister for each run, but vie would expect initialization only once per process. This cl is to teach videocapture android how to deinitialize and allow it to be re-initializable.
braveyao@webrtc.org
2014-05-15 03:18:15 +00:00
54231f0662Ignore the return value of UpdateRtcpTimestamp instead of printing warning. Because UpdateRtcpTimestamp may fail when there's no valid RTCP SR, which can happen in the first couple seconds or when the channel is a send only channel. Either case we don't want the warning log.
wu@webrtc.org
2014-05-14 23:06:23 +00:00
bb6201ae4bTCP remote socket address should have both server hostname and IP address. Hostname is necessary when we are creating TLS based socket, for certificate verification.
mallinath@webrtc.org
2014-05-14 22:43:05 +00:00
a150bc9bbfPeerConnection(android): allow initializing either (or neither) of {Voice,Video}Engine.
fischman@webrtc.org
2014-05-14 22:00:50 +00:00
fischman@webrtc.org