Paul Kerr
|
5809a53505
|
Bug 1027100: visual distortion work-around by re-initializing the vp8 encoder on frame size changes r=jesup
|
2014-06-25 13:40:18 -07:00 |
|
Ehsan Akhgari
|
cbcad1b765
|
Bug 1025393 - Enable building webrtc with clang-cl; r=jesup
--HG--
extra : rebase_source : 16c3846d3a31b71e4ba3f9e4214c1ef8ff6a03e4
|
2014-06-16 18:17:47 -04:00 |
|
Randell Jesup
|
147c58e672
|
Bug 1025176: Save AEC dumps in a specified directory depending on platform/pref r=pkerr
|
2014-06-16 15:51:45 -04:00 |
|
Randell Jesup
|
e3e7209c97
|
Bug 1024288: Allow aec debug data to be dumped on the fly, with max size r=pkerr
|
2014-06-12 12:20:10 -04:00 |
|
Ed Morley
|
a5c42af943
|
Backed out changeset 7b4feb3d3a39 (bug 1024288) for compilation errors; CLOSED TREE
|
2014-06-12 17:41:12 +01:00 |
|
Randell Jesup
|
332ff728b8
|
Bug 1024288: Allow aec debug data to be dumped on the fly, with max size r=pkerr
|
2014-06-12 12:20:10 -04:00 |
|
Randell Jesup
|
3eda6a0803
|
Bug 970713: Adjust webrtc trace buffering for about:webrtc changes r=pkerr
|
2014-06-09 04:34:37 -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
|
12dfa6e7da
|
Bug 951496 - Fix Stastistics typo in vie_codec. r=jesup
|
2014-06-04 23:57:02 -04:00 |
|
Randell Jesup
|
cd089192fd
|
Bug 1003712: Codec availability support and prioritization r=ehugg
|
2014-06-04 14:52:32 -04:00 |
|
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 |
|
Randell Jesup
|
51036cd19e
|
Bug 743703: allow mirroring of trace logs to NSPR; fix backwards lazy allocation defines r=pkerr
|
2014-05-28 03:18:33 -04:00 |
|
Randell Jesup
|
41ccb95961
|
Bug 985254: Add H264 codec-specific structure to carry negotiated data r=pkerr
|
2014-05-24 18:28:01 -04:00 |
|
Randell Jesup
|
fd032ddd4c
|
Bug 985254: Cleaup H264 packetization and jitter buffer r=pkerr
|
2014-05-24 18:28:01 -04:00 |
|
Randell Jesup
|
295343b36e
|
Bug 985254: modify upstream h264 packetization patch to make it work r=pkerr
|
2014-05-24 18:28:01 -04:00 |
|
Randell Jesup
|
fe6e9b77c7
|
Bug 985254: review cleanups from H264 packetization patch r=pkerr
|
2014-05-24 18:28:01 -04:00 |
|
Randell Jesup
|
72962eb159
|
Bug 985254: H264 RTP packetization imported from upstream patchset #10 r=jesup
https://webrtc-codereview.appspot.com/13399004/
|
2014-05-24 18:28:00 -04:00 |
|
Randell Jesup
|
f83fd207cc
|
Bug 981780: fix disable-webrtc r=glandium
|
2014-05-09 14:40:32 -04:00 |
|
Chris Peterson
|
78ae1f032d
|
Bug 1005784 - Fix -Wuninitialized warnings in webrtc/modules/audio_device/linux/. r=jesup
|
2014-05-05 23:38:04 -07:00 |
|
Randell Jesup
|
46a6b9385e
|
Bug 694814: Patch 2: modifications to webrtc.org single_rw_fifo r=glandium,ted
|
2014-04-02 13:58:19 -04:00 |
|
Randell Jesup
|
8fbb6219f2
|
Bug 694814: Patch 1: Add farend input to webrtc.org upstream rs=padenot
|
2014-04-02 13:58:19 -04:00 |
|
Randell Jesup
|
2ade2a2cdc
|
Backed out changeset 89a615263614 (bug 694814)
|
2014-04-07 15:37:55 -04:00 |
|
Randell Jesup
|
373d268aa8
|
Backed out changeset 6922b1261595 (bug 694814)
|
2014-04-07 15:37:54 -04:00 |
|
Randell Jesup
|
29ba637c69
|
Bug 694814: Patch 2: modifications to webrtc.org single_rw_fifo r=glandium,ted
|
2014-04-02 13:58:19 -04:00 |
|
Randell Jesup
|
73e3825d95
|
Bug 694814: Patch 1: Add farend input to webrtc.org upstream rs=padenot
|
2014-04-02 13:58:19 -04:00 |
|
Randell Jesup
|
beb3941cd7
|
Backed out 965c62289427:cb894b5d342f for perma-orange on b2g emulator M10 r=backout
|
2014-04-02 17:11:12 -04:00 |
|
Randell Jesup
|
b3a497d253
|
Bug 694814: Patch 2: modifications to webrtc.org single_rw_fifo r=glandium,ted
|
2014-04-02 13:58:19 -04:00 |
|
Randell Jesup
|
40fe624598
|
Bug 694814: Patch 1: Add farend input to webrtc.org upstream rs=padenot
|
2014-04-02 13:58:19 -04:00 |
|
Jan Beich
|
a0cc624457
|
Bug 985848 - Use videodev2.h on DragonFly/DPorts as well. r=jesup
|
2014-03-24 08:57:58 -04:00 |
|
Kyle Huey
|
510a49016d
|
Bug 967364: Rename already_AddRefed::get to take. r=bsmedberg
|
2014-03-15 12:00:15 -07:00 |
|
Jan-Ivar Bruaroey
|
b31835c861
|
Bug 970686: Undo extensions to wrong rtcp methods in webrtc.org r=jesup
|
2014-03-13 22:28:12 -04:00 |
|
Jan-Ivar Bruaroey
|
57b144b21b
|
Bug 970686: Outbound getStats: Fixed RTCP timestamps and remote packets/bytes received. r=jesup
|
2014-03-14 14:34:02 -04:00 |
|
Gian-Carlo Pascutto
|
5eb3621aee
|
Bug 877954 - Add additional logging for WebRTC adaption & resolution changes. r=jesup
|
2014-03-13 11:06:39 +01:00 |
|
Gian-Carlo Pascutto
|
739c10fdfb
|
Bug 877954 - Enable QM if load adaption is enabled. r=jesup
|
2014-03-13 11:06:27 +01:00 |
|
Gian-Carlo Pascutto
|
fece01ade1
|
Bug 877954 - Push load state to media optimization. Add simple CPU adaption rules. r=jesup
|
2014-03-13 11:05:42 +01:00 |
|
Gian-Carlo Pascutto
|
3fefb76168
|
Bug 877954 - Implement Load Management service. Add callbacks to ViEncoder. r=jesup
|
2014-03-13 11:05:27 +01:00 |
|
Randell Jesup
|
da2a501066
|
Bug 964127: Add logging of webrtc a/v sync status r=jib
|
2014-03-12 20:11:49 -04:00 |
|