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
2386882266Revert "Fix compilation on windows with clang, indentation cleanups"
sergeyu@chromium.org
2014-07-25 17:37:12 +00:00
a44fce5920Fix compilation on windows with clang, indentation cleanups
sergeyu@chromium.org
2014-07-25 17:28:25 +00:00
190d269c0fRefactor 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
2014-07-25 10:32:30 +00:00
06b04ec4abFix a crash in statscollector.cc caused by invoking methods on the worker thread which destroys the Transport.
jiayl@webrtc.org
2014-07-24 20:41:20 +00:00
f9460688a6Make sure padding is sent on the first sending RTP module.
mflodman@webrtc.org
2014-07-24 16:41:25 +00:00
8721f989bfRevert 6766 "Temporarily add a default ctor to StatsReport and make |id| non const. As soon as I've updated the chrome side, I'll revert this cl."
henrike@webrtc.org
2014-07-23 21:38:09 +00:00
0f7328cd6bTemporarily add a default ctor to StatsReport and make |id| non const. As soon as I've updated the chrome side, I'll revert this cl.
tommi@webrtc.org
2014-07-23 16:31:57 +00:00
543e589205Wire up VideoOptions for payload-based padding.
pbos@webrtc.org
2014-07-23 07:01:31 +00:00
efe4b9af49Add VP8 video decoding hw acceleration support to Java Peerconnection library. For now NVidia decoder is supported only, Qualcomm will be added once b/16353967 is fixed.
glaznev@webrtc.org
2014-07-22 17:44:53 +00:00
6f48f1bf68Implement encoder options in WebRtcVideoEngine2.
pbos@webrtc.org
2014-07-22 16:29:54 +00:00
cadd078cf9Remove unused config.h and math.h includes.
pbos@webrtc.org
2014-07-22 15:26:09 +00:00
194fea7640The lastest commit on this file was in
minyue@webrtc.org
2014-07-22 09:55:51 +00:00
ceafa8cce9MIPS optimizations for ISAC (patch #2)
andrew@webrtc.org
2014-07-21 16:43:13 +00:00
908f57ed94Disable GetStatsForInvalidTrack while I rewrite it.
tommi@webrtc.org
2014-07-21 11:44:39 +00:00
756b8462ebRefactor 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
2014-07-21 11:24:17 +00:00
fd61a1d693Revert 6745 "Refactor StatsCollector and associated types." Broke build on android.
tommi@webrtc.org
2014-07-21 11:05:28 +00:00
647e05cfcdRefactor 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
2014-07-21 10:55:11 +00:00
3c10758b3bCheck before send/receive rtp header extensions.
pbos@webrtc.org
2014-07-20 15:27:35 +00:00
8636fc852eCreates the default track if the remote media content is send-only and there is no stream in the SDP.
jiayl@webrtc.org
2014-07-18 20:54:27 +00:00
ff50debd37Runtime guard for iOS7 property.
tkchin@webrtc.org
2014-07-18 17:17:59 +00:00
9343cf67a9Fix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS.
tkchin@webrtc.org
2014-07-18 17:13:28 +00:00
ba92c52570Disable GetStats on DrMemory.
pbos@webrtc.org
2014-07-18 13:33:48 +00:00
e364ac902fAudioBuffer: Optimize const accesses to arrays that autoconvert int16<->float
kwiberg@webrtc.org
2014-07-18 07:50:29 +00:00
c145668dc8Reduce runtime of RingBufferTest by a factor of 100.
andrew@webrtc.org
2014-07-17 23:16:44 +00:00
4f5da030f1Use _numMixedParticipants instead of audioFrameList->size() to determine if there're more than one participants.
wu@webrtc.org
2014-07-17 22:19:21 +00:00
aa93611375Connect to the turn server if address cannot be resolved by the browser by using unresolved address. This case is only considered for TCP sockets. P2P layer will assume socket will do the resolve by using a proxy.
mallinath@webrtc.org
2014-07-17 21:55:04 +00:00
e5995aadd5Assigning a priority to TURN server list passed to PeerConnection. First entry in the TURN server list will get the highest priotity and so forth.
mallinath@webrtc.org
2014-07-17 18:23:52 +00:00
8b94e3da0fFix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled.
stefan@webrtc.org
2014-07-17 16:10:14 +00:00
4065988108Remove unused ExperimentalNS API in AudioProcessing
aluebs@webrtc.org
2014-07-17 11:32:09 +00:00
2b6bc8d84fAudioBuffer: Eliminate the SplitChannelBuffer class
kwiberg@webrtc.org
2014-07-17 09:46:37 +00:00
5301b0f1fcMove additional state into WebRtcVideoSendStream.
pbos@webrtc.org
2014-07-17 08:51:46 +00:00
2561d52460Simplify AudioBuffer::mixed_low_pass_data API
aluebs@webrtc.org
2014-07-17 08:27:39 +00:00
af93fc08a1AudioBuffer: Let ChannelBuffer handle bounds checking of channel parameter
kwiberg@webrtc.org
2014-07-17 08:18:33 +00:00
2ade42bd96Add unit test for MediaFile WAV file writing
kwiberg@webrtc.org
2014-07-17 08:11:32 +00:00
4a472fb18dFixes up rtc so that it compiles on iOS 8 SDK. Adds support for UIInterfaceOrientationUnknown (new with in SDK) and makes it the same as UIInterfaceOrientationPortrait.
tkchin@webrtc.org
2014-07-17 00:21:59 +00:00
52eddec71bRevert 6707 "Add support of multiple STUN servers in UDPPort."
wu@webrtc.org
2014-07-17 00:03:24 +00:00
c56ae63ea6r6709 lacks a change in BUILD.gn
minyue@webrtc.org
2014-07-16 22:18:49 +00:00
74aaf29a0fRaw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.
minyue@webrtc.org
2014-07-16 21:28:26 +00:00
4c3e9917e7Make sure b lines appear before all the a lines. Per RFC 4566, the order of media description should be: m= (media name and transport address) i=* (media title) c=* (connection information -- optional if included at session level) b=* (zero or more bandwidth information lines) k=* (encryption key) a=* (zero or more media attribute lines)
wu@webrtc.org
2014-07-16 21:03:13 +00:00
46fb331bc5Add support of multiple STUN servers in UDPPort. Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.
jiayl@webrtc.org
2014-07-16 20:55:31 +00:00
2e3c97ddf5Compile-time guard for iOS7 specific property.
tkchin@webrtc.org
2014-07-16 19:59:05 +00:00
4070b1db53Print an info log instead of return an error if an external encoder is de-registered, but no corresponding internal encoder can be registered automatically.
stefan@webrtc.org
2014-07-16 11:20:40 +00:00
63c60ed224Remove old padding path in RTPSender.
pbos@webrtc.org
2014-07-16 09:37:29 +00:00
efb81d8d1fint16<->float conversions: Use size_t for array length argument, not int
kwiberg@webrtc.org
2014-07-16 08:36:52 +00:00
0fa6366ed1Define convenient FATAL_ERROR() and FATAL_ERROR_IF() macros
kwiberg@webrtc.org
2014-07-16 08:34:58 +00:00
e8ea33ccb1nrsh1 is written before tmp321 is read, so needs to be earlyclobber
kwiberg@webrtc.org
2014-07-16 08:26:48 +00:00
38ce7d03d8Implement unittest for SetSendCodecsChangesExistingStreams.
pbos@webrtc.org
2014-07-16 08:01:38 +00:00
bac5f0fb56Fix an invalid memory access due to typo in win/cursor.cc.
jiayl@webrtc.org
2014-07-15 20:32:03 +00:00
122caa51b1After an audio interruption the audio unit no longer invokes its render callback, which results in a loss of audio. Restarting the audio unit post interruption fixes the issue.
tkchin@webrtc.org
2014-07-15 20:20:47 +00:00
47218956fcMinor refactoring of StatsCollector. * Make GetTimeNow a static method in the cc file. * Make GetTransportIdFromProxy a static method as well and not a class method.
tommi@webrtc.org
2014-07-15 19:22:37 +00:00
324f63ca38rtc::Fatal output: Print space between # and message
kwiberg@webrtc.org
2014-07-15 11:41:05 +00:00
bc73871251Remove the VPM denoiser.
pbos@webrtc.org
2014-07-15 09:50:40 +00:00
2adc51c86eHandle the case if an unusually long peer name is provided in the peerconnection example.
tommi@webrtc.org
2014-07-15 08:56:07 +00:00
cb859ecd3bReplace strcpy with talk_base::strcpyn.
pbos@webrtc.org
2014-07-15 08:28:20 +00:00
6823479ad3Roll libyuv from 1033 to 1035 to get cpuid fix for AVX2 that avoids misdetect causing a crash in AVX2 code on cpus that do not have AVX2. BUG=libyuv:343 TESTED=libyuv try bots pass R=andrew@webrtc.org
fbarchard@google.com
2014-07-14 23:27:05 +00:00
henrike@webrtc.org