generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Commit Graph

  • 4d14592c67 rtc::Buffer: Restore length method for backwards compatibility kwiberg@webrtc.org 2015-03-24 12:51:58 +00:00
  • deafa7b3c9 Remove I420VideoFrame::SwapFrame magjed@webrtc.org 2015-03-24 12:43:05 +00:00
  • 2d2a30c2e2 Remove I420VideoFrame::CloneFrame magjed@webrtc.org 2015-03-24 12:37:36 +00:00
  • 0b52cebd28 Improve logging and add DCHECKs in codec database. pbos@webrtc.org 2015-03-24 11:20:54 +00:00
  • eebcab5ce9 rtc::Buffer: Rename length to size, for conformance with the STL kwiberg@webrtc.org 2015-03-24 09:19:06 +00:00
  • e815290828 Update README instructions for Android AppRTCDemo. glaznev@webrtc.org 2015-03-23 22:35:14 +00:00
  • a5f6fb53ba Permit single-stream max bitrates above 2000k. pbos@webrtc.org 2015-03-23 22:29:39 +00:00
  • a197a5eed6 Update libsrtp includes in preparation of roll into Chromium. jiayl@webrtc.org 2015-03-23 22:11:49 +00:00
  • a3ffc56cee Allow setting thread priorities in Chromium on all but linux platforms. The previous check was overly broad, so narrowing it down to linux only. tommi@webrtc.org 2015-03-23 20:11:11 +00:00
  • 39fc1d3d48 Disable PeerConnectionClientTest.testLoopbackVp9 henrik.lundin@webrtc.org 2015-03-23 19:57:30 +00:00
  • 0b44b58a3c Limit disabling of PeerConnectionEndToEndTest.Call to Windows henrik.lundin@webrtc.org 2015-03-23 19:47:53 +00:00
  • 64eb2ff0b9 iOS library build script tkchin@webrtc.org 2015-03-23 19:07:37 +00:00
  • 9509fbfc30 Split EventWrapper in twain. I'm splitting the timer functions in EventWrapper into a separate interface. - Users of the timer functions have different needs than users of a generic event - Providing a default implementation for EventWrapper that simply uses rtc::Event. tommi@webrtc.org 2015-03-23 16:25:09 +00:00
  • 82e8ae4ee8 Disable PeerConnectionEndToEndTest.Call in libjingle_peerconnection_unittest henrik.lundin@webrtc.org 2015-03-23 14:25:13 +00:00
  • 2b4ce3a501 Convert webrtc/video/ abort/assert to CHECK/DCHECK. pbos@webrtc.org 2015-03-23 13:12:24 +00:00
  • 41d2befe9f Limit RED audio payload to narrow band. minyue@webrtc.org 2015-03-23 12:57:45 +00:00
  • 1596a4f88b Temporarily disable SetPriority when building with Chromium. This is due to errors we were hitting with Chromium's sandbox policy for pthread_setschedparam. tommi@webrtc.org 2015-03-23 12:39:21 +00:00
  • d4e7d49628 Scaler: Recycle allocations using buffer pool. magjed@webrtc.org 2015-03-23 12:27:55 +00:00
  • 09b6ff9460 Disable PLC for iSAC henrik.lundin@webrtc.org 2015-03-23 12:23:51 +00:00
  • ee0c5af314 Remove unused version.py script. kjellander@webrtc.org 2015-03-23 12:20:00 +00:00
  • aa0bbab8ec Fix build failure jmarusic@webrtc.org 2015-03-23 11:42:45 +00:00
  • a4bef3e6c0 AcmReceiver: use std::map instead of an array to keep the list of decoders jmarusic@webrtc.org 2015-03-23 11:19:35 +00:00
  • 3335a4ffc8 Prevent asserting on unset start bitrate. pbos@webrtc.org 2015-03-23 09:48:41 +00:00
  • 50ed0d9630 Roll chromium_revision 6311617..da9a1c0 (321517:321718) kjellander@webrtc.org 2015-03-23 07:12:50 +00:00
  • e5e92bd556 Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows (fix) kjellander@webrtc.org 2015-03-22 16:27:50 +00:00
  • cfde27eeb3 Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows. kjellander@webrtc.org 2015-03-22 16:09:15 +00:00
  • 38492c5b6f Re-land 8810 "- Add a SetPriority method to ThreadWr..." tommi@webrtc.org 2015-03-22 14:41:46 +00:00
  • 90a1cb4630 Revert 8810 "- Add a SetPriority method to ThreadWrapper" Seeing if this is causing roll issues. tommi@webrtc.org 2015-03-22 14:33:54 +00:00
  • b789f6271a Re-land 8809 "Set WebRtcVideoEngine2 as the WebRtcMe..." tommi@webrtc.org 2015-03-22 12:50:30 +00:00
  • 0c3400168a Revert 8809 "Set WebRtcVideoEngine2 as the WebRtcMediaEngine." content_browsertests started failing around the time the change landed and rolls are failing now. I'm going to try rolling this back, start a roll, and then re-land. tommi@webrtc.org 2015-03-22 12:45:23 +00:00
  • 346a64b9b5 Mac would force bluetooth playout working with 8kHz/1ch if capturing/rendering shares the same device, e.g. changing from 44.1kHz/2ch as default. So in the HandleStreamFormatChange() callback, we need to re-initiate the playout as same as what we do in InitPlayout(). Here we merely copy those codes out from InitPlayout() into a new SetDesiredPlayoutFormat() function for the invoking from the two places. Previously, HandleStreamFormatChange only re-creates the AudioConverter, which is not enough. We also need to reset the buffer size and refresh the latency. braveyao@webrtc.org 2015-03-21 01:05:56 +00:00
  • 4553941d32 Document the 'int' return value of Resampler methods. wtc@chromium.org 2015-03-20 23:28:07 +00:00
  • 3200a64b3c Minor fix for MIPS Android build. andrew@webrtc.org 2015-03-20 22:55:14 +00:00
  • 4ddc9387bd Support VP8 hardware encoding and decoding on IA devices. glaznev@webrtc.org 2015-03-20 21:20:44 +00:00
  • b9557a9bb7 Fix code to handle crashes for non-VP8. pbos@webrtc.org 2015-03-20 19:52:56 +00:00
  • b6817d793f - Add a SetPriority method to ThreadWrapper - Remove 'priority' from CreateThread and related member variables from implementations - Make supplying a name for threads, non-optional tommi@webrtc.org 2015-03-20 15:51:39 +00:00
  • 66df3cf7ab Set WebRtcVideoEngine2 as the WebRtcMediaEngine. pbos@webrtc.org 2015-03-20 15:44:49 +00:00
  • 8296ec518b Fix heap-use-after-free in WebRtcVideoEngine2. pbos@webrtc.org 2015-03-20 14:27:49 +00:00
  • a3209a2b27 Release buffer pool in Vp8DecoderImpl::Release(). pbos@webrtc.org 2015-03-20 13:35:56 +00:00
  • 8904290aca Make screenshare target bitrate experiment always on pbos@webrtc.org 2015-03-20 12:49:38 +00:00
  • d9c5024ee7 Roll chromium_revision bd49b12..6311617 (320783:321517) kjellander@webrtc.org 2015-03-20 12:34:49 +00:00
  • 9f9ea7e5ab Clean up webrtc external capture. This cl removes the dependency to the external capture module if external capturing is used in webrtc. It also removes two external capture methods that is not needed. Further more it adds I420VideoFrame::Create that takes a pointer to packed memory as input. perkj@webrtc.org 2015-03-20 10:55:15 +00:00
  • 443ad403f5 Remove FullStackTest frame pointer handles. pbos@webrtc.org 2015-03-20 07:34:28 +00:00
  • 6231fb6dac Prevent crashes when copying a zero-size frame. pbos@webrtc.org 2015-03-20 07:33:02 +00:00
  • 6069032ebb Refactor audio_coding/isac: removed usage of macro WEBRTC_SPL_LSHIFT_W32 bjornv@webrtc.org 2015-03-20 07:03:28 +00:00
  • 4ab23d0e8f Refactor audio_coding/ilbc: removes usage of macro WEBRTC_SPL_LSHIFT_W32 bjornv@webrtc.org 2015-03-20 06:01:06 +00:00
  • bd8c865f43 Remove build-time beamformer flags. andrew@webrtc.org 2015-03-20 00:28:22 +00:00
  • 04c50981f8 Add the Ooura FFT to RealFourier. andrew@webrtc.org 2015-03-19 20:06:29 +00:00
  • ba86031b34 Whitespace change to trigger new Git pollers (2). kjellander@webrtc.org 2015-03-19 18:10:28 +00:00
  • cf3fb9b3ba Whitespace change to trigger new Git pollers. kjellander@webrtc.org 2015-03-19 17:50:44 +00:00
  • 80d9aeeda5 Adds full-duplex unit test to AudioDeviceTest on Android henrika@webrtc.org 2015-03-19 15:28:16 +00:00
  • 361981faa8 Use scoped_ptr for ThreadWrapper::CreateThread. tommi@webrtc.org 2015-03-19 14:44:18 +00:00
  • c7d5a733b0 Disable flaky test on DrMemory bots tina.legrand@webrtc.org 2015-03-19 14:43:56 +00:00
  • 27c0be9dfe Remove ThreadObj #define and kThreadMaxNameLength from thread_wrapper. tommi@webrtc.org 2015-03-19 14:35:58 +00:00
  • 0c26299739 Disabling two flaky tests in libjingle_media_unittest. tina.legrand@webrtc.org 2015-03-19 13:27:50 +00:00
  • 17c64d1c96 Revert "Put ViEFrameProviderBase::DeliverFrame back in the critical section in ViECapturer::DeliverI420Frame" magjed@webrtc.org 2015-03-19 10:57:35 +00:00
  • c7157da599 Use atomic operations for setting/reading the trace filter. The filter is currently being set and read by a number of threads and tripping up tsan. tommi@webrtc.org 2015-03-19 09:30:29 +00:00
  • 9afaee74ab Reland 8749: AudioEncoder: return EncodedInfo from Encode() and EncodeInternal() jmarusic@webrtc.org 2015-03-19 08:50:26 +00:00
  • d21406d333 Remove command-line tool 'video_coding_test'. pbos@webrtc.org 2015-03-19 08:18:53 +00:00
  • c4709a2930 Split C++ class from macro overrides to fix Chromium build tommi@webrtc.org 2015-03-19 07:25:53 +00:00
  • 5506a93efd Expose ViECaptureImpl::DisconnectCaptureDevice() to JNI of WebRTCDemo and call it before releasing camera to deregister the corresponding framecallback. Also stop camera after stop remote rendering as the correct termination order. braveyao@webrtc.org 2015-03-19 00:12:23 +00:00
  • 8cc47e926c Objective-C readability review. tkchin@webrtc.org 2015-03-18 23:38:04 +00:00
  • 2a8a46dacb vp8: Add missing call to SetUsageMessage(). kjellander@webrtc.org 2015-03-18 21:08:37 +00:00
  • 8f76cd25ec Renaming neteq_opus_fec_quality_test. minyue@webrtc.org 2015-03-18 20:43:40 +00:00
  • 840da7b755 Implement Rotation in Android Renderer. guoweis@webrtc.org 2015-03-18 16:58:13 +00:00
  • 143451d259 Base start bitrate on last observed bitrate. pbos@webrtc.org 2015-03-18 14:40:03 +00:00
  • 5a477a0bc6 DCHECK frame parameters instead of return codes. pbos@webrtc.org 2015-03-18 14:11:39 +00:00
  • 4346d92578 Use SendTimeHistory to keep track of send times in simulations. stefan@webrtc.org 2015-03-18 13:40:54 +00:00
  • f18993323d Removing henrik.lundin from OWNERS in video_coding/* henrik.lundin@webrtc.org 2015-03-18 09:55:59 +00:00
  • af612d5e07 Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."" perkj@webrtc.org 2015-03-18 09:51:05 +00:00
  • 6dba1ebd14 Make AudioDecoder stateless henrik.lundin@webrtc.org 2015-03-18 09:47:08 +00:00
  • 14ee8cc9c7 WebRtcVideoFrame: Support odd resolutions magjed@webrtc.org 2015-03-18 09:21:58 +00:00
  • fc562e0a56 Delete ACMGenericCodec::Encode and use AudioEncoder::Encode directly henrik.lundin@webrtc.org 2015-03-18 07:32:13 +00:00
  • 019955d770 Revert 8749 "We changed Encode() and EncodeInternal() return typ..." tommi@webrtc.org 2015-03-18 06:38:04 +00:00
  • 3fffd66dfa Revert "Implement Rotation in Android Renderer." guoweis@webrtc.org 2015-03-18 04:20:03 +00:00
  • 835ec63d8a Implement Rotation in Android Renderer. guoweis@webrtc.org 2015-03-18 02:44:11 +00:00
  • 52cd828e17 Allow webrtc external encoder factories to declare encoders have internal camera sources. pthatcher@webrtc.org 2015-03-18 02:24:43 +00:00
  • edd517bca1 Fix FYI build - add a missing include to event_tracer.h in system_wrappers. tommi@webrtc.org 2015-03-17 22:14:57 +00:00
  • 54d072ea20 Add CVO support to video_coding layer. guoweis@webrtc.org 2015-03-17 21:54:50 +00:00
  • 63a10978e1 Remove troublesome Windows line ending. pthatcher@webrtc.org 2015-03-17 21:48:39 +00:00
  • 462dbcfc2a Fix bug in Transport where channel_.clear() was being called without a lock. Looks like this snuck in between misaligned braces. tommi@webrtc.org 2015-03-17 21:40:06 +00:00
  • b493cb4497 Add storage alignment fix for opengles2.0 for iOS tkchin@webrtc.org 2015-03-17 20:18:15 +00:00
  • da4fcc494c Add minor fixes to video_capture_ios.mm in order to make it more robust. tkchin@webrtc.org 2015-03-17 20:13:15 +00:00
  • 2161234cf6 Add new features to AppRTCDemo from private repo. glaznev@webrtc.org 2015-03-17 18:23:31 +00:00
  • 779c3d16b9 Use ByteReader/ByteWriter instead of rtputility and manual shift/add. sprang@webrtc.org 2015-03-17 16:42:49 +00:00
  • 09098dabd3 Fix screenshare loopback target bitrate which isn't correctly configured sprang@webrtc.org 2015-03-17 16:27:41 +00:00
  • 25819b8294 Revert 8753 "Use atomic operations for setting/reading the trace..." Caused VP9 test to fail on TSAN and doesn't build in some configuration due to "../webrtc/base/criticalsection.h:181:12: error: cannot compile this atomic library call yet" :-( tommi@webrtc.org 2015-03-17 15:35:11 +00:00
  • b91d0f5130 1. Have IPIsPrivate calling IPIsLinkLocal 2. Also check the Mac based IPv6 3. move the ip filtering into createnetwork. It shouldn't be done in IsIgnoredNetwork as the IP inside that could change later. guoweis@webrtc.org 2015-03-17 14:43:20 +00:00
  • 3093390479 Parsing of transport wide sequence number rtp extension header. Plus some refactoring to correctly handle padding. sprang@webrtc.org 2015-03-17 14:33:12 +00:00
  • 1e6925274a Write commit position as a comment in Chromium DEPS. kjellander@webrtc.org 2015-03-17 14:30:08 +00:00
  • 7c64ed2e0c Move trace_event and associated files to webrtc/base. Also starting to use TRACE_EVENT from thread.cc in webrtc/base, to track Invoke() calls. tommi@webrtc.org 2015-03-17 14:25:37 +00:00
  • 7c112f3e5a Adding build_opus as a switch in GYP. minyue@webrtc.org 2015-03-17 14:04:56 +00:00
  • c383c24c2b Use atomic operations for setting/reading the trace filter. The filter is currently being set and read by a number of threads and tripping up tsan. tommi@webrtc.org 2015-03-17 13:46:42 +00:00
  • a846371ace Modify EventPosix to prevent spurious wakeups. pbos@webrtc.org 2015-03-17 13:11:15 +00:00
  • a78a94e838 Fix RateTracker to set an initial reference time when first updated. perkj@webrtc.org 2015-03-17 12:45:15 +00:00
  • e155dbeae9 VP8/9EncoderImpl::Encode: Check resolution of input I420VideoFrame magjed@webrtc.org 2015-03-17 12:27:26 +00:00
  • 0cb612b43b We changed Encode() and EncodeInternal() return type from bool to void in this issue: https://webrtc-codereview.appspot.com/38279004/ Now we don't have to pass EncodedInfo as output parameter, but can return it instead. This also adds the benefit of making clear that EncodeInternal() needs to fill in this info. jmarusic@webrtc.org 2015-03-17 12:12:17 +00:00
  • 73d763e71f Add I420 buffer pool to avoid unnecessary allocations magjed@webrtc.org 2015-03-17 11:40:45 +00:00
  • ae222b5be6 Remove dead code in WebRtcVideoEngine2 unittests. pbos@webrtc.org 2015-03-17 10:47:42 +00:00
  • 858024f1d9 WebRtcVideoFrame: Initialize members in empty constructor magjed@webrtc.org 2015-03-17 08:46:54 +00:00