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192,172 Commits 1 Branch 165 Tags 1.2 GiB
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261 Commits

Author SHA1 Message Date
Randell Jesup
5b5bb47836 Bug 989944: Increase decode timestamp map to handle delayed decode on 8x10 r=jesup 2014-07-03 12:46:28 -04:00
Randell Jesup
a7c962bbd9 Bug 979716: Make Opus complexity configurable in WebRTC; default Gonk to complexity 1 r=jmspeex 2014-07-01 05:10:44 -04:00
Gian-Carlo Pascutto
403e14b8d6 Bug 1018928 - Work around Camera focus mode bug in some Android devices. r=blassey 2014-06-27 12:13:50 +02:00
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