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1942 Commits

Author SHA1 Message Date
Karl Tomlinson
e58f9c45b1 b=1023697 use MediaStream to convert between stream time and seconds/ticks in MediaPipeline r=roc 
The fake graph needs an implementation of the conversion methods.

The real graph will change to use audio ticks for time in a subsequent patch,
but the fake graph doesn't know about MEDIA_TIME_FRAC_BITS, so that change
can be made now in the fake graph.

--HG--
extra : transplant_source : %22%C4%01Yh%5D%F0%A6%11%40%CD%B5%89%0A%8C%8A%C2%19%5E%CC
 2014-06-12 16:44:58 +12:00
EKR
a2a4a9a1bc Bug 1022812 - Link a debuggable version of NSS into media/mtransport. r=mt 2014-06-11 07:17:02 -07:00
Chris Peterson
ce766e4253 Bug 1023075 - Fix more clang warnings in webrtc/signaling. r=jesup 2014-06-09 22:42:11 -07:00
Randell Jesup
3eda6a0803 Bug 970713: Adjust webrtc trace buffering for about:webrtc changes r=pkerr 2014-06-09 04:34:37 -04:00
Jan-Ivar Bruaroey
8b459224fd Bug 970713 - Add 'Start Debug Mode' button to about:webrtc. r=smaug, r=Unfocused, r=jesup 2014-06-08 21:00:12 -04:00
Paul Kerr [:pkerr]
af0b5dd5d3 Bug 970713 - Part 1: Control webrtc logging from about:config settings r=jesup 2014-06-08 18:54:47 -07:00
Anthony Jones
fa4301f766 Bug 1016150 - Fix Windows date assert in libstagefright demuxer; r=cpearce 2014-06-09 18:07:46 +12:00
Randell Jesup
370f28d765 Bug 999704: Implement GMP codec interface to webrtc (not enabled yet) r=joshmoz,ehugg,jesup,pkerr 2014-06-08 17:25:18 -04:00
Ryan VanderMeulen
0ae54304d5 Backed out changeset 2af237fa2079 (bug 999704) for bustage. 
CLOSED TREE DONTBUILD
 2014-06-08 14:39:44 -04:00
Randell Jesup
8cf755ddd9 Bug 999704: Implement GMP codec interface to webrtc (not enabled yet) r=joshmoz,ehugg,jesup 2014-06-08 14:07:53 -04:00
Randell Jesup
442154b7cb Bug 970742: Add receive state monitoring to webrtc CodecStatistics r=jib 2014-06-08 11:06:30 -04:00
Randell Jesup
fc5f6c61d2 Bug 970742: Monitor decoder error state to enable recording errors and error recovery times r=jib 2014-06-08 10:33:02 -04:00
Jan-Ivar Bruaroey
73c28df208 Bug 951496 - Codec telemetry. r=jesup 2014-06-07 17:33:39 -04:00
Jan-Ivar Bruaroey
f23107dd2f Bug 951496 - Codec getStats. r=smaug, r=jesup 2014-06-07 17:27:26 -04:00
Steven Lee
96d69b8623 Bug 951496 - Statistics data for checking the status of codec. r=jesup 2014-06-04 23:56:30 -04:00
Jan-Ivar Bruaroey
12dfa6e7da Bug 951496 - Fix Stastistics typo in vie_codec. r=jesup 2014-06-04 23:57:02 -04:00
Byron Campen [:bwc]
0a653f0e9c Bug 1004530 - Part 3: Unit test that verifies that new pairs will start when local gather happens after all preceding pairs have failed, provided the grace period has not elapsed. Also a couple more tests that use a new test-case feature. 2014-06-04 17:21:59 -07:00
Byron Campen [:bwc]
582d112d7e Bug 1004530 - Part 2: Unit test for verifying that local candidates gathered after the check timer has stopped are ultimately scheduled. 2014-06-03 10:56:54 -07:00
Byron Campen [:bwc]
e28d5a8b95 Bug 1004530 - Part 1: Allow a grace period for trickle candidates to arrive when all candidate pairs have failed. r=drno, r=ekr 2014-05-01 14:07:54 -07:00
Adam Roach [:abr]
df82c8e1e7 Bug 1018372 - Check aThread against mThread in PeerConnectionImpl constructor r=jesup 2014-06-06 15:56:47 -05:00
Karl Tomlinson
0b9ed65c05 b=1015828 match Fake_MediaStreamListener::NotifyPull time advances to timer period and Fake_AudioStreamSource::Periodic buffer size r=rjesup 
Also, increment Fake_SourceMediaStream::mDesiredTime each period,
instead of each listener notification.

--HG--
extra : rebase_source : 723a2a3b126adca486154d0b686746c21dbac37e
 2014-06-05 10:11:51 +12:00
Wes Kocher
f6bae13ecb Merge m-c to inbound on a CLOSED TREE 2014-06-04 18:48:20 -07:00
Star Cheng
cc24600c70 Bug 1007552 - To support publicnotification audio channel type for camera shutter. r=kinetik 2014-05-22 15:08:05 +08:00
Jacek Caban
716717fb10 Bug 1018905 - Fix media/libstagefright compilation on mingw. r=ajones 
--HG--
extra : rebase_source : 9f89d48a4985bda9611ff39b90955d026e173808
 2014-06-03 13:26:07 +02:00
Chris Peterson
1c51032add Bug 1017110 - Suppress warnings in third-party code: libstagefright. r=cajbir 
--HG--
extra : rebase_source : 7bb135a1ac1cdeda748fcfb6a2a6283807259e80
 2014-05-26 22:31:34 -07:00
Byron Campen [:bwc]
bbaf4386c7 Bug 998989 - Part 1: ChromeOnly API for getting notifications when PCs are initted, or change ICE connection/gathering state. Also, expose the PC id, and allow getAllStats to be filtered by the same. r=jib, r=bz 2014-05-22 14:14:56 -07:00
Robert O'Callahan
a8bbe633b9 Bug 1015664. Part 2: Remove some NS_HIDDEN usage. r=bsmedberg 2014-06-03 00:08:24 +12:00
EKR
4884fdde56 Bug 1018473. Unit test for duplicate trickle candidates. r=bwc 2014-05-31 12:06:45 -07:00
Byron Campen [:bwc]
7b0fa364cc Bug 1018473: Detect when vcmRxAllocICE has already been called for a given stream, and suppress the (duplicate) connection to SignalCandidate. r=ekr 2014-05-31 19:41:53 -07:00
Adam Roach [:abr]
04d763f9a3 Bug 1017755 - Make DTLS 'would have blocked' messages less aggressive r=jesup 2014-05-30 20:02:36 -05:00
Randell Jesup
cd089192fd Bug 1003712: Codec availability support and prioritization r=ehugg 2014-06-04 14:52:32 -04:00
Randell Jesup
7e84082c49 Bug 1003712: Use Media Resource Manager to reserve OMX Codecs r=jhlin 2014-06-04 14:52:31 -04:00
Byron Campen [:bwc]
1774026f94 Bug 1017291 - Keep track of the number of errors in AddIceCandidate before ICE completes, and record this number in telemetry in the success and failure cases separately. r=ekr 2014-05-29 08:40:31 -07:00
Mike Hommey
41657ceb81 Fix non-unified build bustage from bug 987979 on a CLOSED TREE. r=me 2014-05-30 09:32:08 +09:00
Randell Jesup
5aaae2b64e Bug 987979: Patch 12 - Add webrtc JNI target annotations to stop ProGuard from removing too much code. r=blassey 2014-05-29 17:05:16 -04:00
Randell Jesup
d65b42fede Bug 987979: Patch 11 - Add webrtc 3.50 support for Froyo/Gingerbread/Ice Cream Sandwich. r=blassey 2014-05-29 17:05:16 -04:00
Randell Jesup
5b9598c2f6 Bug 987979: Patch 10 - Support building with older Android SDKs. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
8cb8e97704 Bug 987979: Patch 9 - Use Android JNI Wrappers for off-thread FindClass and Global Context. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
12d756308d Bug 987979: Patch 8 - Support rotating/suspending/resuming an ongoing WebRTC call. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
a8d21229d7 Bug 987979: Patch 7 - Remove JSON/UCI requirements for Camera capture capability. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
e2805a0c2d Bug 987979: Patch 6 - Include CPU feature detection source directly. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
500b3d6ff7 Bug 987979: Patch 5 - Enable switching between OpenSLES and JNI backends, dummy OpenSLES output. r=rjesup 2014-05-29 17:05:14 -04:00
Randell Jesup
4465789496 Bug 987979: Patch 4 - Rework WebRTC.org audio code for Mozilla integration. r=jesup 2014-05-29 17:05:14 -04:00
Randell Jesup
79df25773b Bug 987979: Patch 3 - Fix various build issues in webrtc.org/Mozilla integration. r=rjesup 2014-05-29 17:05:14 -04:00
Randell Jesup
7740e2ceb2 Bug 987979: Patch 2 - Rollup of changes previously applied to media/webrtc/trunk/webrtc rs=jesup 2014-05-29 17:05:14 -04:00
Randell Jesup
66465cce72 Bug 987979: Patch 1 - Webrtc updated to branch 3.50 rev 5764, pull made Mon Mar 24 15:36:34 EDT 2014 rs=jesup 
--HG--
rename : media/webrtc/trunk/webrtc/video_engine/new_include/config.h => media/webrtc/trunk/webrtc/config.h
rename : media/webrtc/trunk/webrtc/video_engine/new_include/frame_callback.h => media/webrtc/trunk/webrtc/frame_callback.h
rename : media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRTCAudioDevice.java => media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
rename : media/webrtc/trunk/webrtc/common_unittest.cc => media/webrtc/trunk/webrtc/test/common_unittest.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/direct_transport.h => media/webrtc/trunk/webrtc/test/direct_transport.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/fake_decoder.cc => media/webrtc/trunk/webrtc/test/fake_decoder.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/fake_decoder.h => media/webrtc/trunk/webrtc/test/fake_decoder.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/fake_encoder.cc => media/webrtc/trunk/webrtc/test/fake_encoder.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/fake_encoder.h => media/webrtc/trunk/webrtc/test/fake_encoder.h
rename : media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/fake_network_pipe_unittest.cc => media/webrtc/trunk/webrtc/test/fake_network_pipe_unittest.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/flags.cc => media/webrtc/trunk/webrtc/test/flags.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/flags.h => media/webrtc/trunk/webrtc/test/flags.h
rename : media/webrtc/trunk/webrtc/common_video/test/frame_generator.h => media/webrtc/trunk/webrtc/test/frame_generator.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/frame_generator_capturer.cc => media/webrtc/trunk/webrtc/test/frame_generator_capturer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/frame_generator_capturer.h => media/webrtc/trunk/webrtc/test/frame_generator_capturer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/gl/gl_renderer.cc => media/webrtc/trunk/webrtc/test/gl/gl_renderer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/gl/gl_renderer.h => media/webrtc/trunk/webrtc/test/gl/gl_renderer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/linux/glx_renderer.cc => media/webrtc/trunk/webrtc/test/linux/glx_renderer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/linux/glx_renderer.h => media/webrtc/trunk/webrtc/test/linux/glx_renderer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/linux/video_renderer_linux.cc => media/webrtc/trunk/webrtc/test/linux/video_renderer_linux.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/mac/run_tests.mm => media/webrtc/trunk/webrtc/test/mac/run_tests.mm
rename : media/webrtc/trunk/webrtc/video_engine/test/common/mac/video_renderer_mac.h => media/webrtc/trunk/webrtc/test/mac/video_renderer_mac.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/mac/video_renderer_mac.mm => media/webrtc/trunk/webrtc/test/mac/video_renderer_mac.mm
rename : media/webrtc/trunk/webrtc/video_engine/test/common/null_platform_renderer.cc => media/webrtc/trunk/webrtc/test/null_platform_renderer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/null_transport.cc => media/webrtc/trunk/webrtc/test/null_transport.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/null_transport.h => media/webrtc/trunk/webrtc/test/null_transport.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/rtp_rtcp_observer.h => media/webrtc/trunk/webrtc/test/rtp_rtcp_observer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/run_loop.h => media/webrtc/trunk/webrtc/test/run_loop.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/run_tests.h => media/webrtc/trunk/webrtc/test/run_tests.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/statistics.cc => media/webrtc/trunk/webrtc/test/statistics.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/statistics.h => media/webrtc/trunk/webrtc/test/statistics.h
rename : media/webrtc/trunk/webrtc/video_engine/test/test_main.cc => media/webrtc/trunk/webrtc/test/test_main.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/vcm_capturer.cc => media/webrtc/trunk/webrtc/test/vcm_capturer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/vcm_capturer.h => media/webrtc/trunk/webrtc/test/vcm_capturer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/video_capturer.cc => media/webrtc/trunk/webrtc/test/video_capturer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/video_capturer.h => media/webrtc/trunk/webrtc/test/video_capturer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/video_renderer.cc => media/webrtc/trunk/webrtc/test/video_renderer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/video_renderer.h => media/webrtc/trunk/webrtc/test/video_renderer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/win/d3d_renderer.cc => media/webrtc/trunk/webrtc/test/win/d3d_renderer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/win/d3d_renderer.h => media/webrtc/trunk/webrtc/test/win/d3d_renderer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/win/run_loop_win.cc => media/webrtc/trunk/webrtc/test/win/run_loop_win.cc
rename : media/webrtc/trunk/webrtc/video_engine/new_include/transport.h => media/webrtc/trunk/webrtc/transport.h
rename : media/webrtc/trunk/webrtc/video_engine/test/loopback.cc => media/webrtc/trunk/webrtc/video/loopback.cc
rename : media/webrtc/trunk/webrtc/video_engine/internal/transport_adapter.cc => media/webrtc/trunk/webrtc/video/transport_adapter.cc
rename : media/webrtc/trunk/webrtc/video_engine/internal/transport_adapter.h => media/webrtc/trunk/webrtc/video/transport_adapter.h
rename : media/webrtc/trunk/webrtc/video_engine/new_include/video_renderer.h => media/webrtc/trunk/webrtc/video_renderer.h
 2014-05-29 17:05:13 -04:00
Nils Ohlmeier [:drno]
5abd2eec9b Bug 1014304 - Remove onconnection and onclosedconnection from RTCPeerConnection. r=jib, r=jesup, r=smaug 2014-05-28 09:36:00 -04:00
Jan-Ivar Bruaroey
d30b322032 Bug 859565 - Remove legacy PeerConnectionImpl.readyState. r=bholley, r=abr 2014-05-17 17:11:27 -04:00
Byron Campen [:bwc]
caa58710f6 Bug 1016724 - Make sure the word "gathering" appears in the timecard stamp for complete. r=jesup 2014-05-27 17:19:45 -07:00
Byron Campen [:bwc]
865d18fb03 Bug 891551 - Part 2: (Upliftable) Fix bugs where PR_WOULD_BLOCK_ERROR (or, in some cases, PR_NOT_CONNECTED_ERROR while a TCP socket was connecting) would cause sockets to be abandoned for no good reason (see also bug 985493 and 1001671). r=bwc 2014-05-02 10:49:00 -07:00