Commit Graph

2018 Commits

Author SHA1 Message Date
Anthony Jones
70f58351d1 Bug 1020679 - Fix MP4 demuxer duration. r=cpearce 2014-06-06 16:19:29 +12:00
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
d49880509b Bug 1025349: fix error in ccsnap line label indexes r=ehugg 2014-06-16 15:10:16 -04:00
Randell Jesup
beda527078 Bug 1025354: fix out-of-sync name array for SIPCC logs r=ehugg 2014-06-16 15:10:05 -04:00
JW Wang
c0e8ee9fea Bug 1008079 - Add cubeb_resampler.cpp to build files. r=glandium 2014-06-16 19:46:00 -04:00
JW Wang
221afb22dc 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
6536a96fff 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
636b64db51 Bug 1024510 - Unbreak --with-system-nss build after bug 1022812. r=mshal 2014-06-16 01:02:25 -04:00
Randell Jesup
d96b743305 Bug 1025343: fix issues with overlong codec names in AudioConduit r=pkerr 2014-06-16 01:00:33 -04:00
Randell Jesup
0bedd46970 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
95ddaacac2 Bug 1022235: Make the webrtc LoadManager/LoadMonitor a singleton r=bsmedberg,pkerr 2014-06-13 15:50:51 -04:00
Byron Campen [:bwc]
4a0a0e70ca 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
1cf15ce508 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
7db6a9b53c 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
07a0927480 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
9ef43aa872 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
9d1bc6e5a6 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
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
Ed Morley
03812e8f9a Backed out changeset 5d63a1316981 (bug 1024288) 2014-06-12 17:40:44 +01:00
Randell Jesup
0a434412d2 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
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
7aefb73722 Bug 1017332: log WebRTC SDP parse errors due to no \n r=ehugg 2014-06-12 12:03:42 -04:00
Byron Campen [:bwc]
1f7eeb76d8 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]
881184c858 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
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
Carsten "Tomcat" Book
9c73951370 merge b2g-inbound to mozilla-central 2014-05-28 14:33:48 +02:00
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
Anthony Jones
fc378f051a Bug 1014814 - Fix Android log r=glandium 2014-05-26 09:20:56 +12:00
Enda Mannion
2a2092dc76 Bug 1003994 - H.246 and multiple video codec tests. r=jesup 2014-05-26 10:07:19 +01:00
Jan-Ivar Bruaroey
93980709ca Bug 970685, telemetry for WebRTC bandwidth, stats-tweak approach. r=jesup 2014-05-27 14:41:17 -04:00
Jan-Ivar Bruaroey
723947759f Bug 970685 - tweak internal RTCStatsQuery to use nsAutoPtr for report, so it can be stolen 2014-05-27 12:58:03 -04:00
EKR
df92fcf432 Bug 1015409 - Fix trickle between CreateOffer() and SetLocal(). r=bwc 2014-05-27 13:13:43 -07:00
Jan Beich
6966cac66c Bug 1014613 followup - Add one more fix for OpenBSD. 2014-05-26 16:05:53 +12:00
Anthony Jones
1bfb005447 Bug 1014626 - Fix Windows 64 build break 2014-05-26 15:52:01 +12:00
Randell Jesup
3e5d95d980 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
c102072fa1 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
388a6314ad Bug 1014921: Wallpaper 8x10 OMX H264 encode/decode mismatch by forcing IDRs r=jhlin 2014-05-24 18:28:03 -04:00
Randell Jesup
ff9cdaf529 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
d8de28f2bb 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
bc26a0f3d5 Bug 1015029: Use OMX_VIDEO_ControlRateConstantSkipFrames mode for H.264 OMX encoder r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
6143419992 Bug 989945: add a bit more logging to H264 OMX codec r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
9216b39c92 Bug 989945: Use configureDirect to set OMX HW H264 encoder config correctly r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
06ca7ee4bf Bug 989945: increase logging for H264 OMX code r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
88dd15fc1e Bug 985253: Support H.264 RTP mode 1 support in webrtc signaling r=ehugg 2014-05-24 18:28:02 -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