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998 Commits

Author SHA1 Message Date
Randell Jesup
a02f87eea0 Bug 1003712: Codec availability support and prioritization r=ehugg 2014-06-04 14:52:32 -04:00
Randell Jesup
e324737c53 Bug 1003712: Use Media Resource Manager to reserve OMX Codecs r=jhlin 2014-06-04 14:52:31 -04:00
Byron Campen [:bwc]
74c49d3d46 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
2a92625af7 Bug 1015664. Part 2: Remove some NS_HIDDEN usage. r=bsmedberg 2014-06-03 00:08:24 +12:00
EKR
1ea7cf9b40 Bug 1018473. Unit test for duplicate trickle candidates. r=bwc 2014-05-31 12:06:45 -07:00
Byron Campen [:bwc]
01ccd3683c 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
Byron Campen [:bwc]
3d1bd46584 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
bcfae34d17 Fix non-unified build bustage from bug 987979 on a CLOSED TREE. r=me 2014-05-30 09:32:08 +09:00
Randell Jesup
b5ac06a0e7 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
9f738ae94f 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
7d91d878c8 Bug 987979: Patch 10 - Support building with older Android SDKs. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
07bd430f23 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
3b05d7cae8 Bug 987979: Patch 8 - Support rotating/suspending/resuming an ongoing WebRTC call. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
4812c3eb03 Bug 987979: Patch 7 - Remove JSON/UCI requirements for Camera capture capability. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
66ce6ff1ad Bug 987979: Patch 6 - Include CPU feature detection source directly. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
2f742c010a 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
5c562e73d6 Bug 987979: Patch 4 - Rework WebRTC.org audio code for Mozilla integration. r=jesup 2014-05-29 17:05:14 -04:00
Randell Jesup
964601c191 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
21318d2311 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
0654f9ad2b 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]
24c7206622 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
16cde275f8 Bug 859565 - Remove legacy PeerConnectionImpl.readyState. r=bholley, r=abr 2014-05-17 17:11:27 -04:00
Byron Campen [:bwc]
6e411ef3aa Bug 1016724 - Make sure the word "gathering" appears in the timecard stamp for complete. r=jesup 2014-05-27 17:19:45 -07:00
Randell Jesup
3e5a393123 Bug 743703: allow mirroring of trace logs to NSPR; fix backwards lazy allocation defines r=pkerr 2014-05-28 03:18:33 -04:00
Enda Mannion
944f1623f0 Bug 1003994 - H.246 and multiple video codec tests. r=jesup 2014-05-26 10:07:19 +01:00
Jan-Ivar Bruaroey
fd6c40896d Bug 970685, telemetry for WebRTC bandwidth, stats-tweak approach. r=jesup 2014-05-27 14:41:17 -04:00
Jan-Ivar Bruaroey
68e827df7d Bug 970685 - tweak internal RTCStatsQuery to use nsAutoPtr for report, so it can be stolen 2014-05-27 12:58:03 -04:00
EKR
017a4114a2 Bug 1015409 - Fix trickle between CreateOffer() and SetLocal(). r=bwc 2014-05-27 13:13:43 -07:00
Randell Jesup
f4bb1be0ae 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
0f38f52f24 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
2dc6a27a54 Bug 1014921: Wallpaper 8x10 OMX H264 encode/decode mismatch by forcing IDRs r=jhlin 2014-05-24 18:28:03 -04:00
Randell Jesup
d6733b246b 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
8370108c2b 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
05874298f3 Bug 1015029: Use OMX_VIDEO_ControlRateConstantSkipFrames mode for H.264 OMX encoder r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
28f4f67da7 Bug 989945: add a bit more logging to H264 OMX codec r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
3c43eb9170 Bug 989945: Use configureDirect to set OMX HW H264 encoder config correctly r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
3cb67e15cc Bug 989945: increase logging for H264 OMX code r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
6ddd2f12d7 Bug 985253: Support H.264 RTP mode 1 support in webrtc signaling r=ehugg 2014-05-24 18:28:02 -04:00
Randell Jesup
f8592cfe05 Bug 985254: Add H264 codec-specific structure to carry negotiated data r=pkerr 2014-05-24 18:28:01 -04:00
Randell Jesup
146015cbf8 Bug 985254: Cleaup H264 packetization and jitter buffer r=pkerr 2014-05-24 18:28:01 -04:00
Randell Jesup
fef0238f25 Bug 985254: modify upstream h264 packetization patch to make it work r=pkerr 2014-05-24 18:28:01 -04:00
Randell Jesup
4a993a5cdc Bug 985254: review cleanups from H264 packetization patch r=pkerr 2014-05-24 18:28:01 -04:00
Randell Jesup
80e5a12d40 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
5d71bfa862 Bug 1004396: Make video codec default bitrates configurable for WebRTC r=ekr 2014-05-24 18:28:00 -04:00
Kyle Huey
8c5cca136c Bug 996133: Remove unnecessary NS_DISPATCH_NORMAL arguments to NS_DispatchToMainThread. r=ehsan 2014-05-23 12:53:17 -07:00
Jan-Ivar Bruaroey
98ea0e204d Bug 1013238 - Fix timer event crash on shutdown in recent PeerConnectionCtx change. r=jesup 2014-05-21 22:32:03 -04:00
Anders Lund
efc993ca04 Bug 942188 - Added parsing of ice-lite attribute and start ice checks as controlling if peer is ice-lite. r=abr 2014-05-16 01:32:00 -05:00
Byron Campen [:bwc]
743284e6c7 Bug 1013729 - Null check in case PushLayers failed when registering for the DTLS connection signal. r=jesup 2014-05-21 08:49:03 -07:00
Carsten "Tomcat" Book
a79c1ec7ff Backed out changeset 9b2588d41e3a (bug 969395) for bustage 2014-05-21 11:29:21 +02:00
Qiang Lu
c9bd29e3a3 Bug 969395 - Add stub library for accesing VP8 HW codec through android native mediacodec interface. r=rjesup 2014-05-21 10:14:31 +08:00
EKR
b9c43f7c18 Bug 1012999: When STUN global rate limit is exceeded, record this in telemetry. r=ekr 2014-05-19 19:16:38 -07:00
Jan-Ivar Bruaroey
5a14073109 Bug 970685 - Thread approach for WebRTC telemetry for jitter, packet-loss and RTT. r=jesup 2014-05-10 08:54:41 -04:00
John Lin
32e9769fb9 Bug 1011422 - Clear mOMXConfigured flag to correctly restart OMX H.264 encoder. r=jesup 2014-05-18 19:30:00 +02:00
Carsten "Tomcat" Book
fef547ad4e Backed out changeset 426b187eae45 (bug 1001422) wrong bugnumber in commit message 2014-05-19 11:44:00 +02:00
John Lin
3ba9c2eabc Bug 1001422 - Clear mOMXConfigured flag to correctly restart OMX H.264 encoder. r=jesup 2014-05-18 19:30:00 +02:00
John Lin
33b91e228f Bug 1010841 - Handle on-demand key frame request in OMX H.264 encoder. r=jesup 2014-05-16 01:56:00 -04:00
Randell Jesup
f884ae7641 Bug 1011214: Release OMX monitor when shutting down Encoder output drain thread r=jhlin 2014-05-16 04:37:08 -04:00
Randell Jesup
71910ce4dc Bug 981780: fix disable-webrtc r=glandium 2014-05-09 14:40:32 -04:00
Martin Thomson
5acda6875e Bug 966066 - Add principal observer to RTCPeerConnection. r=jib 2014-04-25 10:34:00 -04:00
Neil Rashbrook
5b3f3e053a Bug 514280 Only use nsCOMPtr for interfaces r=bsmedberg 2014-05-11 10:47:11 +01:00
Ryan VanderMeulen
893e7c7ecb Backed out changeset 047f98eef5cf (bug 1007196) for intermittent failures. 2014-05-09 14:13:21 -04:00
Ethan Hugg
6b867e1cb5 Bug 1007196 - Re-enable the Signaling unittests for Linux and OSX. r=ted 2014-05-07 13:04:34 -07:00
Chris Peterson
475d4e8367 Bug 990764 - Replace MOZ_ASSUME_UNREACHABLE in webrtc/signaling. r=jesup 2014-04-19 11:05:10 -07:00
Neil Rashbrook
fac8c73779 Backout of bug 514280 changeset c738f7348dea for build failure on a CLOSED TREE 2014-05-08 20:35:09 +01:00
Neil Rashbrook
5b1f7b4a77 Bug 514280 Only use nsCOMPtr for interfaces r=bsmedberg 2014-05-08 20:08:38 +01:00
Chris Peterson
b65637a65e Bug 1005784 - Fix -Wuninitialized warnings in webrtc/modules/audio_device/linux/. r=jesup 2014-05-05 23:38:04 -07:00
Byron Campen [:bwc]
2c84d5f0bc Bug 1002831 - Display remote and local SDP on about:webrtc. r=smaug, r=jib 2014-05-05 11:13:24 -07:00
Byron Campen [:bwc]
a622f4666a Bug 970734 - Part 2: Record final ICE/media stats when PeerConnections are closed, so they show up in about:webrtc. r=smaug, r=jib 2014-05-05 09:35:57 -07:00
Robert O'Callahan
e1bb2e935f Bug 1006248. Part 4: Use better #include paths for libstagefright headers in a couple of places. r=glandium 
--HG--
extra : rebase_source : e8c7e019b0bc5bf60081aad158a7d89fbb261e29
 2014-05-06 17:40:59 +12:00
Martin Thomson
5471fc97e1 Bug 1006112 - Fixing regressions in signaling_unittests. r=ekr 2014-05-05 14:19:00 +02:00
Martin Thomson
c3c2709899 Bug 942367 - Stream isolation for WebRTC r=bholley 2014-05-01 12:51:00 +02:00
Ethan Hugg
3ae788c1d5 Bug 1002890 - Signaling unittests no longer need exceptions to mainthread checks. r=jesup 2014-04-28 19:45:40 -07:00
Ethan Hugg
2e714cf592 Bug 819549 - Signaling unittests should dispatch to main thread when calling PC. r=jesup 2014-04-28 15:00:19 -07:00
Randell Jesup
95437d211a Bug 985253: Send rtcp-fb for all video codecs, and fix answer generation for H.264 for rtcp-fb r=ehugg 2014-04-30 18:18:35 -04:00
John Lin
7e66846d66 Bug 1002402: typo fix for adjusting SPS/PPS timestamps r=jesup 2014-04-30 01:20:41 -04:00
John Lin
4fd3ee35e5 Bug 1002402: (temporary) change SPS/PPS timestamps so webrtc jitter buffer won't drop them r=jesup 2014-04-29 13:25:40 -04:00
Ed Morley
e41a3e1c8a Merge mozilla-central and inbound 2014-04-29 18:23:29 +01:00
Randell Jesup
4cfc4d0e45 Bug 1002306: don't accidentally disable non-H264 codecs in the OMX code r=ekr 2014-04-28 19:52:16 -04:00
John Lin
b9eab230af Bug 911046 - Get graphic buffers of decoded frames through gonk native window callback. r=jesup 2014-04-27 21:07:00 -04:00
John Lin
4bf83010dc Bug 1002402 - Support RTP H.264 input data in WebRTC OMX decoder. r=jesup 2014-04-28 23:37:00 +02:00
Byron Campen [:bwc]
ad61a0ec58 Bug 1001959 - Give up references to NrIceMediaStream on STS instead of main. r=jib 2014-04-28 09:01:29 -07:00
Birunthan Mohanathas
5f1fde8824 Bug 900908 - Part 3: Change uses of numbered macros in nsIClassInfoImpl.h/nsISupportsImpl.h to the variadic variants. r=froydnj 2014-04-27 03:06:00 -04:00
Garvan Keeley
40aeac4872 Bug 1001708: Don't use ternary operator with class const statics r=jesup 2014-04-27 00:02:17 -04:00
Byron Campen [:bwc]
799148596a Bug 970690 - Part 2: Add basic telemetry for ICE. r=mt 2014-03-03 10:58:35 -08:00
Martin Thomson
6c6647cf1a Bug 1001539 - Fix compilation warning in ccsip_pmh.c. r=bwc 2014-04-25 10:58:00 -04:00
Paul Kerr [:pkerr]
8dc11ae04a Bug 970691 - Part 2: Restore digit stamping function to YuvStamper. r=jesup 
Refactor digit writing method to use the new internals. Allows digit string
to wrap through multiple lines in a small frame.
 2014-04-24 19:58:21 -07:00
Paul Kerr [:pkerr]
07fe6406b7 Bug 970691 - Part 1: Add timestamp to fake video. r=jesup 
Update YuvStamper utility. Add a CRC32 to the encoded
payload and have the decode method us this to verify reception.
Wrap encoded values across multiple lines in the frame buffer
when necessary. Use YuvStamper to encode a timestamp in each fake video frame.
Extract the value in VideoConduit to calculate the video latency
and add this to a running average latency when enabled via config.
 2014-03-22 16:35:43 -07:00
John
356b84852a Bug 999902 - Enable WebRTC OMX codec only when Android version >= 18. r=jesup 2014-04-23 02:59:00 +02:00
Wes Kocher
50c5a26e84 Backed out 2 changesets (bug 970691) for build bustage on a CLOSED TREE 
Backed out changeset 83f7aec5a083 (bug 970691)
Backed out changeset 94348d189ed5 (bug 970691)
 2014-04-23 18:26:05 -07:00
Paul Kerr [:pkerr]
c45ebab36f Bug 970691 - Part2: Restore digit stamping function to YuvStamper. r=jesup 
Refactor digit writing method to use the new internals. Allows digit string
to wrap through multiple lines in a small frame.
 2014-04-23 10:03:18 -07:00
Paul Kerr [:pkerr]
7db94a6847 Bug 970691 - Part 1: Add timestamp to fake video. r=jesup 
Update YuvStamper utility. Add a CRC32 to the encoded
payload and have the decode method us this to verify reception.
Wrap encoded values across multiple lines in the frame buffer
when necessary. Use YuvStamper to encode a timestamp in each fake video frame.
Extract the value in VideoConduit to calculate the video latency
and add this to a running average latency when enabled via config.
 2014-03-22 16:35:43 -07:00
John Lin
14da65f440 Bug 911046 - Part 6: Make H.264 preferred video codec when enabled in preferences. r=jesup, ekr 2014-04-21 23:44:00 +02:00
John Lin
9153e591e1 Bug 911046 - Part 5: Register H.264 external codec for B2G. r=jesup, ekr 2014-04-21 23:43:00 +02:00
John Lin
2a88e07bb8 Bug 911046 - Part 4: Add external H.264 HW video codec implementation for B2G. r=jesup 2014-04-21 23:42:00 +02:00
John Lin
704708f61b Bug 911046 - Part 2: Support 'handle-using' video frames for WebRTC on B2G. r=jesup, ekr 2014-04-21 23:41:00 +02:00
John Lin
7df65a157b Bug 911046 - Part 1: Support external codec in VideoConduit. r=jesup 2014-04-21 23:40:00 +02:00
Ethan Hugg
95043ad5a4 Bug 995380 - Signaling unittests should use the real main thread. r=jesup 2014-04-21 19:37:22 -07:00
Ryan VanderMeulen
ecb85b74fb Backed out changesets 1e581e74878d, 7d2138e87ca0, and 7cc66aee4341 (bug 942367) for B2G mochitest failures. 
CLOSED TREE
 2014-04-17 22:26:07 -04:00
Randell Jesup
dd18e038e2 Bug 996853: handle AUDIO_FORMAT_SILENCE in MediaPipeline and AudioSegment::WriteTo r=roc 2014-04-17 17:45:25 -04:00
Martin Thomson
33ae3b1f29 Bug 942367 - Part 3: Stream isolation for WebRTC. r=jib, r=bholley 2014-04-10 11:52:08 -07:00