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2018 Commits

Author SHA1 Message Date
Anthony Jones
777b028f90 Bug 1020679 - Fix MP4 demuxer duration. r=cpearce 2014-06-06 16:19:29 +12:00
Ehsan Akhgari
6df6805088 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
d4746d1de5 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
735438f351 Bug 1025349: fix error in ccsnap line label indexes r=ehugg 2014-06-16 15:10:16 -04:00
Randell Jesup
c2b8cdaaff Bug 1025354: fix out-of-sync name array for SIPCC logs r=ehugg 2014-06-16 15:10:05 -04:00
JW Wang
fc92b87533 Bug 1008079 - Add cubeb_resampler.cpp to build files. r=glandium 2014-06-16 19:46:00 -04:00
JW Wang
73a1053383 Bug 1008079 - Use a resampler when the sample rate is not supported by the audio engine in cubeb_opensl.c. r=padenot 2014-06-16 19:45:00 -04:00
JW Wang
2ff8979857 Bug 1008079 - Extract the resampling code from cubeb_wasapi.cpp so it is reusable. r=padenot 2014-06-16 19:27:00 -04:00
Jan Beich
9c80ad52ff Bug 1024510 - Unbreak --with-system-nss build after bug 1022812. r=mshal 2014-06-16 01:02:25 -04:00
Randell Jesup
06a8193337 Bug 1025343: fix issues with overlong codec names in AudioConduit r=pkerr 2014-06-16 01:00:33 -04:00
Randell Jesup
f398cfeee3 Bug 1025106: if someone passes us a bogus videocodec config, say it's 'unknown' r=pkerr 2014-06-16 01:00:25 -04:00
Randell Jesup
e3cb840cc2 Bug 1022235: Make the webrtc LoadManager/LoadMonitor a singleton r=bsmedberg,pkerr 2014-06-13 15:50:51 -04:00
Byron Campen [:bwc]
24e05c32c0 Bug 1017332 - Part 1: Dump the r_log ringbuffer on all ICE failures. r=ekr, r=jesup 2014-06-12 17:22:00 -07:00
Mike Hommey
d96ad9d710 Bug 1024260 - Fixup dependencies in media/libopus/Makefile.in to avoid celt_pitch_xcorr_arm-gnu.o being always rebuilt. r=ted,r=me 2014-06-13 10:05:26 +09:00
Wes Kocher
508e00dfab Back out two changesets (bug 1024260) for android build failures on a CLOSED TREE 
* * *
Backed out changeset 9d92de0ada7d (bug 1024260)
* * *
Backed out changeset 5264e512b53c (bug 1024260)
 2014-06-12 17:41:25 -07:00
Mike Hommey
f11f6a062b Fixup for bug 1024260 because mkdir_deps can only be used after including rules.mk. r=me 2014-06-13 09:19:16 +09:00
Mike Hommey
03438d234a Bug 1024260 - Fixup dependencies in media/libopus/Makefile.in to avoid celt_pitch_xcorr_arm-gnu.o being always rebuilt. r=ted 2014-06-13 08:44:48 +09:00
Randell Jesup
9ecc4f0aa3 Bug 1024288: Add a button to about:webrtc to turn on/off AEC logging r=jib,smaug,unfocused 2014-06-12 12:21:38 -04:00
Randell Jesup
a004aeb3ad 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
ea9936f870 Backed out changeset 7b4feb3d3a39 (bug 1024288) for compilation errors; CLOSED TREE 2014-06-12 17:41:12 +01:00
Ed Morley
866f114249 Backed out changeset 5d63a1316981 (bug 1024288) 2014-06-12 17:40:44 +01:00
Randell Jesup
3668de4ea0 Bug 1024288: Add a button to about:webrtc to turn on/off AEC logging r=jib,smaug,unfocused 2014-06-12 12:21:38 -04:00
Randell Jesup
2e622d9a43 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
f2d53111fa Bug 1017332: log WebRTC SDP parse errors due to no \n r=ehugg 2014-06-12 12:03:42 -04:00
Byron Campen [:bwc]
4765de2c12 Bug 1008796 - Fix return value in nr_ice_component_stun_server_default_cb. r=ekr 2014-06-10 10:45:01 -07:00
Byron Campen [:bwc]
0bcb4baac2 Bug 1022776 - Bump max transmit count by 1 and modify unit-tests to compensate. r=ekr 2014-06-09 17:31:44 -07:00
Karl Tomlinson
b1de31746b 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
cc9ec6c21f Bug 1022812 - Link a debuggable version of NSS into media/mtransport. r=mt 2014-06-11 07:17:02 -07:00
Chris Peterson
688dce7e4b Bug 1023075 - Fix more clang warnings in webrtc/signaling. r=jesup 2014-06-09 22:42:11 -07:00
Randell Jesup
f7c6fe9f6e Bug 970713: Adjust webrtc trace buffering for about:webrtc changes r=pkerr 2014-06-09 04:34:37 -04:00
Jan-Ivar Bruaroey
fdfec6e2f2 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]
6cf4c86116 Bug 970713 - Part 1: Control webrtc logging from about:config settings r=jesup 2014-06-08 18:54:47 -07:00
Anthony Jones
b5c97017c7 Bug 1016150 - Fix Windows date assert in libstagefright demuxer; r=cpearce 2014-06-09 18:07:46 +12:00
Randell Jesup
2f70ec93bb 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
e58e907b46 Backed out changeset 2af237fa2079 (bug 999704) for bustage. 
CLOSED TREE DONTBUILD
 2014-06-08 14:39:44 -04:00
Randell Jesup
fe44092940 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
2234ee97f5 Bug 970742: Add receive state monitoring to webrtc CodecStatistics r=jib 2014-06-08 11:06:30 -04:00
Randell Jesup
cffbb35214 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
b7ca6e1f78 Bug 951496 - Codec telemetry. r=jesup 2014-06-07 17:33:39 -04:00
Jan-Ivar Bruaroey
c8f9921b5d Bug 951496 - Codec getStats. r=smaug, r=jesup 2014-06-07 17:27:26 -04:00
Steven Lee
2151e81cd4 Bug 951496 - Statistics data for checking the status of codec. r=jesup 2014-06-04 23:56:30 -04:00
Jan-Ivar Bruaroey
8f90844f7f Bug 951496 - Fix Stastistics typo in vie_codec. r=jesup 2014-06-04 23:57:02 -04:00
Byron Campen [:bwc]
1682c22a73 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]
674bbdc927 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]
f7bb1b3ccf 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]
561bcea3c7 Bug 1018372 - Check aThread against mThread in PeerConnectionImpl constructor r=jesup 2014-06-06 15:56:47 -05:00
Karl Tomlinson
7093899b67 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
40a1340af8 Merge m-c to inbound on a CLOSED TREE 2014-06-04 18:48:20 -07:00
Star Cheng
6a58609092 Bug 1007552 - To support publicnotification audio channel type for camera shutter. r=kinetik 2014-05-22 15:08:05 +08:00
Jacek Caban
f0142ead6c 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
aec0c25259 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]
625491e04d 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
b3cf47b176 Bug 1015664. Part 2: Remove some NS_HIDDEN usage. r=bsmedberg 2014-06-03 00:08:24 +12:00
EKR
7afad1293c Bug 1018473. Unit test for duplicate trickle candidates. r=bwc 2014-05-31 12:06:45 -07:00
Byron Campen [:bwc]
baaf6d91b7 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]
32166eb344 Bug 1017755 - Make DTLS 'would have blocked' messages less aggressive r=jesup 2014-05-30 20:02:36 -05:00
Randell Jesup
a520439607 Bug 1003712: Codec availability support and prioritization r=ehugg 2014-06-04 14:52:32 -04:00
Randell Jesup
0e3d803a91 Bug 1003712: Use Media Resource Manager to reserve OMX Codecs r=jhlin 2014-06-04 14:52:31 -04:00
Byron Campen [:bwc]
d55fb640f7 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
008e551458 Fix non-unified build bustage from bug 987979 on a CLOSED TREE. r=me 2014-05-30 09:32:08 +09:00
Randell Jesup
b7df766f19 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
16ac9901ab 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
e9ebd1968f Bug 987979: Patch 10 - Support building with older Android SDKs. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
19725fda62 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
2e0efa00cb Bug 987979: Patch 8 - Support rotating/suspending/resuming an ongoing WebRTC call. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
709b079b16 Bug 987979: Patch 7 - Remove JSON/UCI requirements for Camera capture capability. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
13eecb30d5 Bug 987979: Patch 6 - Include CPU feature detection source directly. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
e4b0ab08c4 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
91823edc5a Bug 987979: Patch 4 - Rework WebRTC.org audio code for Mozilla integration. r=jesup 2014-05-29 17:05:14 -04:00
Randell Jesup
8d6c68d8b4 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
720bb9708f 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
a9bd67143a 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]
272b02723f 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
f97999ac92 Bug 859565 - Remove legacy PeerConnectionImpl.readyState. r=bholley, r=abr 2014-05-17 17:11:27 -04:00
Byron Campen [:bwc]
45eef204e2 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]
40a32d4dfd 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
Carsten "Tomcat" Book
b01cb5d079 merge b2g-inbound to mozilla-central 2014-05-28 14:33:48 +02:00
Randell Jesup
917bc3029a Bug 743703: allow mirroring of trace logs to NSPR; fix backwards lazy allocation defines r=pkerr 2014-05-28 03:18:33 -04:00
Anthony Jones
6232d67a0f Bug 1014814 - Fix Android log r=glandium 2014-05-26 09:20:56 +12:00
Enda Mannion
1358b1a4d9 Bug 1003994 - H.246 and multiple video codec tests. r=jesup 2014-05-26 10:07:19 +01:00
Jan-Ivar Bruaroey
efd6320e71 Bug 970685, telemetry for WebRTC bandwidth, stats-tweak approach. r=jesup 2014-05-27 14:41:17 -04:00
Jan-Ivar Bruaroey
566c711ce3 Bug 970685 - tweak internal RTCStatsQuery to use nsAutoPtr for report, so it can be stolen 2014-05-27 12:58:03 -04:00
EKR
9466d32c0f Bug 1015409 - Fix trickle between CreateOffer() and SetLocal(). r=bwc 2014-05-27 13:13:43 -07:00
Jan Beich
2b2f779021 Bug 1014613 followup - Add one more fix for OpenBSD. 2014-05-26 16:05:53 +12:00
Anthony Jones
fefce21291 Bug 1014626 - Fix Windows 64 build break 2014-05-26 15:52:01 +12:00
Randell Jesup
e7d5265c71 Bug 1014819: Replace OMX GetCodecConfig with straight caching of H.264 SPS/PPS r=jhlin 2014-05-24 18:28:03 -04:00
Randell Jesup
57bee8ccbd Bug 985254: Modify H264 OMX code to deal with upstream code inserting start codes r=jhlin 2014-05-24 18:28:03 -04:00
Randell Jesup
8b638bb2f9 Bug 1014921: Wallpaper 8x10 OMX H264 encode/decode mismatch by forcing IDRs r=jhlin 2014-05-24 18:28:03 -04:00
Randell Jesup
bf32c93ac4 Bug 997567: Send iframes for HW H264 encoder when bitrate changes with long GOP r=jhlin 2014-05-24 18:28:03 -04:00
Randell Jesup
0caf937427 Bug 997567: Support reconfiguration for frame-rate changes on OMX H.264 encoder r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
6b97454c34 Bug 1015029: Use OMX_VIDEO_ControlRateConstantSkipFrames mode for H.264 OMX encoder r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
d05373629a Bug 989945: add a bit more logging to H264 OMX codec r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
4d72b31af3 Bug 989945: Use configureDirect to set OMX HW H264 encoder config correctly r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
e0e20d85df Bug 989945: increase logging for H264 OMX code r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
053f28defd Bug 985253: Support H.264 RTP mode 1 support in webrtc signaling r=ehugg 2014-05-24 18:28:02 -04:00
Randell Jesup
7d77e6dcd4 Bug 985254: Add H264 codec-specific structure to carry negotiated data r=pkerr 2014-05-24 18:28:01 -04:00
Randell Jesup
603d2f171c Bug 985254: Cleaup H264 packetization and jitter buffer r=pkerr 2014-05-24 18:28:01 -04:00
Randell Jesup
b4fc97f262 Bug 985254: modify upstream h264 packetization patch to make it work r=pkerr 2014-05-24 18:28:01 -04:00
Randell Jesup
9c6dc827f1 Bug 985254: review cleanups from H264 packetization patch r=pkerr 2014-05-24 18:28:01 -04:00
Randell Jesup
c43277ee0f 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