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ad62c8db72
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648 Commits

Author SHA1 Message Date
Byron Campen [:bwc]
ad62c8db72 Bug 933841. Add event handler to dump RLogRingBuffer on test failure, and clear RLogRingBuffer on test start. r=ekr 2013-11-01 13:50:49 -07:00
Randell Jesup
43f13f1f8e Bug 938070: Fix misplaced #ifdef for GONK in webrtc audio_device_impl from 3.43 merge r=jesup 2013-11-15 11:33:18 -05:00
Nathan Froyd
9829841e7c Bug 933320 - part 1 - make find_sdk.py silently comply if we're not running on a Mac host; r=ted 2013-10-31 13:34:02 -04:00
Brad Lassey
7ee547334b bug 936549 - Tab sharing capture device won't stream, add rgb image support to media pipeline r=jesup 2013-11-10 16:24:37 -05:00
Gian-Carlo Pascutto
62c5789f79 Bug 932112: Add a non-ARM MemoryBarrier function. r=glandium 2013-11-07 20:07:48 -05:00
Gian-Carlo Pascutto
0430169a6c Bug 932112: Initialize both JNI and OpenSLES so fallback can work. r=jesup 2013-11-07 20:07:48 -05:00
Randell Jesup
8c4c3f9d55 Bug 932112: JB reflect for low-latency params r=mfinkle 2013-11-07 20:07:47 -05:00
Gian-Carlo Pascutto
a830153b02 Bug 932112: Use the non-main-thread FindClass implementation r=blassey 2013-11-07 20:07:47 -05:00
Randell Jesup
52771d4abf Bug 932112: Rollup of changes previously applied to media/webrtc/trunk/webrtc rs=jesup 
* * *
* * *
Add AndroidAudioManager to the moz.build files.
 2013-11-07 20:07:47 -05:00
Randell Jesup
836b549082 Bug 932112: Webrtc updated to 5041, pull made Mon Oct 28 12:17:00 EDT 2013 rs=jesup 
--HG--
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/source/acm_common_defs.h => media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
rename : media/webrtc/trunk/webrtc/modules/audio_device/android/org/webrtc/voiceengine/WebRTCAudioDevice.java => media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRTCAudioDevice.java
rename : media/webrtc/trunk/webrtc/modules/audio_processing/test/unit_test.cc => media/webrtc/trunk/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h
rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/receiver_fec_unittest.cc => media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/video_capture/android/java/org/webrtc/videoengine/CaptureCapabilityAndroid.java => media/webrtc/trunk/webrtc/modules/video_capture/android/java/src/org/webrtc/videoengine/CaptureCapabilityAndroid.java
rename : media/webrtc/trunk/webrtc/modules/video_capture/android/java/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java => media/webrtc/trunk/webrtc/modules/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/stream_generator.cc => media/webrtc/trunk/webrtc/modules/video_coding/main/source/test/stream_generator.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/stream_generator.h => media/webrtc/trunk/webrtc/modules/video_coding/main/source/test/stream_generator.h
rename : media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/unit_test.cc => media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/unit_test.h => media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h
rename : media/webrtc/trunk/webrtc/modules/video_render/android/java/org/webrtc/videoengine/ViEAndroidGLES20.java => media/webrtc/trunk/webrtc/modules/video_render/android/java/src/org/webrtc/videoengine/ViEAndroidGLES20.java
rename : media/webrtc/trunk/webrtc/modules/video_render/android/java/org/webrtc/videoengine/ViERenderer.java => media/webrtc/trunk/webrtc/modules/video_render/android/java/src/org/webrtc/videoengine/ViERenderer.java
rename : media/webrtc/trunk/webrtc/modules/video_render/android/java/org/webrtc/videoengine/ViESurfaceRenderer.java => media/webrtc/trunk/webrtc/modules/video_render/android/java/src/org/webrtc/videoengine/ViESurfaceRenderer.java
 2013-11-07 20:07:47 -05:00
Byron Campen [:bwc]
c4e9698bbd Bug 936031 - Attempted fix. r=ehugg 2013-11-07 15:03:06 -08:00
Byron Campen [:bwc]
70ded53d09 Bug 936031 - Test case for bug. r=abr 2013-11-07 14:48:43 -08:00
Carsten "Tomcat" Book
f10da167db merge b2g-inbound to mozilla-central 2013-11-04 13:52:18 +01:00
Chris Pearce
e8d0d063b3 Bug 933579 - Export IsVideoContentType() to VideoUtils, so that it can be used elsewhere, and move all of VideoUtils into namespace mozilla. r=kinetik 2013-11-04 11:45:19 +13:00
Matthew Gregan
1c1e844b44 Bug 837563 - Enable libcubeb's PulseAudio backend. r=glandium 2013-10-31 11:37:28 +13:00
Byron Campen [:bwc]
5cc4d97f6f Bug 906990 - Part 8: Create a chrome-only stats interface, and only expose the candidate pair stats there. r=jib 2013-10-29 10:29:43 -07:00
Byron Campen [:bwc]
e1af51f0e7 Bug 906990 - Part 7: Populate candidate pairs in RTCStatsReport. r=jib 2013-10-28 16:02:00 -07:00
Wes Kocher
39d8f615ff Backed out changeset 00f838879263 (bug 906990) 2013-11-01 17:14:59 -07:00
Wes Kocher
73ea42404f Backed out changeset 57a7a785a964 (bug 906990) 2013-11-01 17:14:54 -07:00
Byron Campen [:bwc]
b65db06520 Bug 906990 - Part 8: Create a chrome-only stats interface, and only expose the candidate pair stats there. r=jib 2013-10-29 10:29:43 -07:00
Byron Campen [:bwc]
64e352b730 Bug 906990 - Part 7: Populate candidate pairs in RTCStatsReport. r=jib 2013-10-28 16:02:00 -07:00
Randell Jesup
cd3e87b2df Bug 932215 - Lazily allocate log buffers for webrtc (4MB saving). r=jib 2013-10-30 12:13:07 -04:00
Nathan Froyd
3879d8a3ee Bug 933071 - add --with-macos-private-frameworks to support cross-compiling; r=mshal 2013-10-31 09:50:26 -04:00
Ethan Hugg
f0456f6520 Bug 901560 - Backout of compatibility-breaking datachannel ice component fix r=jesup 2013-10-29 08:52:04 -07:00
Gregory Szorc
9ada834d56 Bug 927837 - Don't manage generated files in configure; r=glandium 
--HG--
extra : rebase_source : b502ce209de6a0ae10e130644e424687e4fae85e
 2013-10-23 14:43:32 -07:00
Phil Ringnalda
11e739f7c6 Merge m-c to m-i 2013-10-27 19:25:15 -07:00
Jaroslav Kopecký
8dd49a58dd Bug 931590 - Pass proper directory when building --with-system-nspr r=jesup 2013-10-27 19:43:04 -04:00
Nicholas Nethercote
ba1e9bce90 Bug 925584 - Remove some unnecessary jsapi.h inclusions from .cpp files. r=Ms2ger. 
--HG--
extra : rebase_source : 41fcb0e922a519ef679c1c1b6293c2b638e83a48
 2013-10-10 15:22:35 -07:00
Phil Ringnalda
a1f80ad10b Back out f872d288480b:9b86b4e60b29 (bug 929513) for failing to build on Windows 
CLOSED TREE
 2013-10-27 15:38:40 -07:00
David Zbarsky
8ce46e2762 Bug 929513 Part 3: Use some LayerIntSize in gfx/layers r=nical 2013-10-27 17:53:27 -04:00
David Zbarsky
b28c18df90 Bug 929513 Part 1: Use gfx::IntSize for image layer sizes r=nical 2013-10-27 17:53:26 -04:00
Peter Van der Beken
cbf7a0c800 Bug 918345 - Turn on WebIDL binding generation for Window and hook it up to quickstubs. r=bz. 
--HG--
extra : rebase_source : 7bde7ddfe297e189ffa678ca1d9c34000bc904ec
 2013-10-08 17:51:42 +02:00
Ms2ger
34f7a76bb1 Backout changeset 2e466ccc7bd0 for devtools test failures. 2013-10-26 17:02:20 +02:00
Peter Van der Beken
a521d7eace Bug 918345 - Turn on WebIDL binding generation for Window and hook it up to quickstubs. r=bz. 
--HG--
extra : rebase_source : 673c08ef093339e6bfb1418366af5cc5fabe7c4d
 2013-10-08 17:51:42 +02:00
Randell Jesup
f551a29a6a Bug 920325: ntohl() isn't defined on Windows unless you include winsock/winsock2.h r=tbsaunde 2013-10-25 20:46:35 -04:00
Randell Jesup
ee8e35ca44 Bug 920325: Add WebRTC latency logging from capture to RTP and from RTP to speakers r=padenot 2013-10-25 18:13:42 -04:00
Randell Jesup
345ac3892d backout 5f38b1bd3358 for bustage CLOSED TREE 2013-10-25 19:25:54 -04:00
Randell Jesup
60b12a2e89 Bug 930603: Ensure AEC known delay doesn't go negative (rev 4886 at webrtc.org) r=jib 2013-10-25 18:21:33 -04:00
Randell Jesup
8777f9a0f5 Bug 930603: Increase WebRTC AEC tail from 48ms to 128ms (rev 4837 at webrtc.org) r=jib 2013-10-25 18:21:23 -04:00
Randell Jesup
2e3491f74c Bug 920325: Add WebRTC latency logging from capture to RTP and from RTP to speakers r=padenot 2013-10-25 18:13:42 -04:00
Jan-Ivar Bruaroey
de7feceb63 Bug 929534 r=jesup 2013-10-25 10:52:17 -04:00
EKR
595e343db4 Bug 930651 - Increase logging level for periodic A/V counters. r= abr 2013-10-24 19:23:24 -07:00
Patrick Wang
4afedddb6b Bug 881761 - Part 3: Initialize NSS when initializing PeerConnection. r=ekr,bsmith 2013-07-22 10:16:13 +08:00
Patrick Wang
6fd870fe12 Bug 932881: Explicitly cast currentSipccState to uint32_t. r=abr 2013-10-30 18:00:28 +08:00
Brian O'Keefe
4c98f61956 Bug 928709 - Convert chromium-config.mk to mozbuild, r=mshal 2013-10-02 13:17:55 -04:00
Randell Jesup
3ad7a7e6d1 Bug 864654: cleanup AudioConduit r=ekr 2013-10-23 06:20:55 -04:00
Randell Jesup
50c7d4e360 Bug 864654: merge backend for send and receive VideoConduits to match AudioConduits & cleanup r=ekr 2013-10-23 06:20:54 -04:00
Mark Banner
7433db8bcc Bug 920991 - Default stun server ip address should be changed to a domain name. r=abr 2013-10-23 09:59:37 +01:00
Ethan Hugg
fdecaf06b3 Bug 925896 - Signaling - Addref when adding sessiondata_t to hash r=abr 2013-10-22 13:14:43 -07:00
Ehsan Akhgari
3b751c0f92 Bug 928712 - Remove the rest of the unneeded prtypes.h inclusions; rs=bsmedberg 2013-10-20 22:59:48 -04:00