Commit Graph

47 Commits

Author SHA1 Message Date
Randell Jesup
3c5f8d49a1 Backed out 965c62289427:cb894b5d342f for perma-orange on b2g emulator M10 r=backout 2014-04-02 17:11:12 -04:00
Randell Jesup
7a728568ca Bug 985714: Monitor AEC echo levels (ERLE/etc) in gUM r=jib 2014-04-02 13:58:20 -04:00
Randell Jesup
54427b9318 Bug 694814: Patch 5 - Move AEC from PeerConnection to getUserMedia rs=padenot 2014-04-02 13:58:19 -04:00
Randell Jesup
7024ecf09f Bug 694814: Patch 4 - Add audio playout delay config var r=padenot 2014-04-02 13:58:19 -04:00
Randell Jesup
f42c0492df Bug 694814: Patch 3 - Add far-end mixer observer and insert far-end audio for AEC r=padenot 2014-04-02 13:58:19 -04:00
Paul Adenot
cf8bdb40d2 Bug 982490 - Ensure for MSG cycle that each MediaStream write the same number of frames to their AudioStream. r=jesup,roc 2014-03-24 11:06:06 +01:00
Randell Jesup
deda7f6382 Bug 980096: fix leaks of VoiceEngines by reinstating use of ScopedCustomReleasePtr r=khuey
--HG--
rename : media/webrtc/signaling/src/media-conduit/MediaEngineWrapper.h => media/webrtc/signaling/src/common/MediaEngineWrapper.h
2014-03-09 00:18:50 -05:00
Steven Lee
a875f4819c Bug 926746 - Part 2: nsContentPermissionHelper set grant information to GonkPermission. r=jesup 2013-12-05 09:29:07 -05:00
Ehsan Akhgari
58e806c093 Bug 939584 - Build some of the directories under content/media in unified mode; r=roc 2013-11-17 21:31:46 -05:00
Randell Jesup
933821ecb9 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
1af5c4fe0b backout 5f38b1bd3358 for bustage CLOSED TREE 2013-10-25 19:25:54 -04:00
Randell Jesup
a0692a49f2 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
Steven Lee
c36ccf3171 Bug 918056 - Return errors when the mic is occupied. r=rjesup 2013-10-01 08:06:57 -04:00
Randell Jesup
f587d1f5aa Bug 901583: Reapply mozilla patches on top of webrtc.org 3.34, use NEON detection rs=jesup
--HG--
rename : media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_opensles_android.cc => media/webrtc/trunk/webrtc/modules/audio_device/audio_device_opensles.cc
rename : media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_opensles_android.h => media/webrtc/trunk/webrtc/modules/audio_device/audio_device_opensles.h
2013-08-30 02:08:57 -04:00
Joshua Cranmer
46c5cc932a Bug 884061 - Part 3d: Use NS_DECL_THREADSAFE_ISUPPORTS in content/, r=smaug
--HG--
extra : rebase_source : ee869e0ec710259b1f3d1a328bff27c5d2960ea1
2013-07-18 21:21:19 -05:00
Randell Jesup
3e9f97631e Bug 848401: switch from ReentrantMonitors and PR_Lock/Condvar to Monitor. Fix {picture:true} for desktop r=khuey,dao 2013-03-13 11:42:18 -04:00
Randell Jesup
97f75463f6 Bug 842715: Refactor gUM prefs use to be on mainthread, and prepare for constraints r=derf 2013-03-04 16:02:17 -05:00
Daniel Holbert
c43defdf94 Bug 843929 - Part 2: Add '(void) mEchoCancel' to silence Wunused-private-field warnings, until the code that uses it is turned on. r=jesup 2013-02-22 06:59:29 -05:00
Robert O'Callahan
ee2ea45fed Bug 830707. Part 3: Don't constrain AudioSegment to a fixed number of channels. r=jesup
--HG--
extra : rebase_source : feacede00821b6673ce04c886a9c3727a4989404
2013-01-21 09:44:44 +13:00
Randell Jesup
439ab021c2 Bug 818670: Enable AEC in PeerConnection, AGC/NoiseSuppression in gUM (w/bustage fix) r=derf 2013-01-29 11:55:09 -05:00
Ed Morley
b8cab0271d Backout 40f09f7bc670 & fc262e3c635f (bug 818670) for frequent fedora64 mochitest-3 leaks on a CLOSED TREE 2013-01-30 10:32:11 +00:00
Randell Jesup
31ade0db43 Bug 818670: Enable AEC in PeerConnection, AGC/NoiseSuppression in gUM (w/bustage fix) r=derf 2013-01-29 11:55:09 -05:00
Ed Morley
67d41c1674 Backout df75a87cce60 & 19e164f7d88d (bug 818670) for build bustage on a CLOSED TREE 2013-01-29 17:28:30 +00:00
Randell Jesup
eb99b87929 Bug 818670: Enable AEC in PeerConnection, AGC/NoiseSuppression in gUM r=derf 2013-01-29 11:55:09 -05:00
Randell Jesup
da209aea3d Bug 831427: Gate RemoveListener(stream) to avoid calling if Destroy() pending r=roc 2013-01-17 02:38:21 -05:00
Robert O'Callahan
f2ee4df1a0 Bug 827537. Refactor AudioChunk to support having separate buffers for each channel. r=jesup
--HG--
extra : rebase_source : 0aa26e1c3181d9fe5158520d4b33248bae0fa5d0
2012-11-22 18:04:27 +13:00
Randell Jesup
49e05b2a18 Bug 828828: Use monitor around all accesses to stream array r=derf 2013-01-10 11:52:53 -05:00
Randell Jesup
90c9997e1f Bug 816664: wallpaper patch for negative delta times for video frames r=roc,derf 2012-11-30 03:08:17 -05:00
Randell Jesup
42b9a000b3 Bug 811757: Allow the user to explicitly share devices between tabs r=anant 2012-12-31 18:12:12 -05:00
Josh Matthews
24b5b26242 Backed out changeset 7d6255d1c547 (bug 816664) 2012-12-28 19:03:48 -05:00
Randell Jesup
39a4dffd71 Bug 816664: wallpaper patch for negative delta times for video frames r=derf 2012-12-28 15:34:01 -05:00
Randell Jesup
f3cbf201a0 Bug 802399: set capture index for audio input when device is Allocate()ed, plug VoEHw leak r=derf 2012-11-15 18:23:39 -05:00
Randell Jesup
f01b2fce9c Bug 811695: disable internal socket transports for getUserMedia Audio capture r=derf 2012-11-15 17:58:40 -05:00
Nathan Froyd
f9b2a74082 Bug 806618 - rewrite PR_NewLogModule calls to not generate static initializers; r=ehsan 2012-10-29 19:32:10 -04:00
Robert O'Callahan
a0d739c463 Bug 805254. Part 7: Move SampleFormat to mozilla::AudioSampleFormat in its own file. r=kinetik 2012-10-25 23:09:40 +13:00
Randell Jesup
97d57fb78e Bug 803799: Start gUM streams in Success callback; add MediaManager mutex r=anant,roc 2012-10-24 19:21:15 -04:00
Ehsan Akhgari
28c105ccf5 Backed out changeset ea436c6f7d2d (bug 803799), landed on a CLOSED TREE 2012-10-24 20:30:08 -04:00
Randell Jesup
781f490022 Bug 803799: Start gUM streams in Success callback; add MediaManager mutex r=anant,roc 2012-10-24 19:21:15 -04:00
Randell Jesup
ca11404fd3 Bug 802661: Clean up getUserMedia MediaStream handling r=roc,anant 2012-10-17 17:40:14 -04:00
Randell Jesup
2614d8891d Bug 801843: Change how video frames are inserted into getUserMedia streams to remove blocking r=roc,anant 2012-10-17 05:46:40 -04:00
Anant Narayanan
ba618700c7 Bug 802411: Refactor MediaEngine to use GIPS singletons; r=jesup 2012-10-16 17:53:55 -07:00
Randell Jesup
d1d511ba1f Bug 773649: Support getting audio and video in the same getUserMedia call r=roc,anant 2012-10-15 16:41:46 -04:00
Paul Adenot
d6650a6ddf Bug 783953 - Rename MOZ_SAMPLE_TYPE_S16LE to MOZ_SAMPLE_TYPE_S16. r=kinetik,roc 2012-09-01 11:35:56 -04:00
Ehsan Akhgari
8c296bbcd4 Bug 579517 - Part 1: Automated conversion of NSPR numeric types to stdint types in Gecko; r=bsmedberg
This patch was generated by a script.  Here's the source of the script for
future reference:

function convert() {
echo "Converting $1 to $2..."
find . ! -wholename "*nsprpub*" \
       ! -wholename "*security/nss*" \
       ! -wholename "*/.hg*" \
       ! -wholename "obj-ff-dbg*" \
       ! -name nsXPCOMCID.h \
       ! -name prtypes.h \
         -type f \
      \( -iname "*.cpp" \
         -o -iname "*.h" \
         -o -iname "*.c" \
         -o -iname "*.cc" \
         -o -iname "*.idl" \
         -o -iname "*.ipdl" \
         -o -iname "*.ipdlh" \
         -o -iname "*.mm" \) | \
    xargs -n 1 sed -i -e "s/\b$1\b/$2/g"
}

convert PRInt8 int8_t
convert PRUint8 uint8_t
convert PRInt16 int16_t
convert PRUint16 uint16_t
convert PRInt32 int32_t
convert PRUint32 uint32_t
convert PRInt64 int64_t
convert PRUint64 uint64_t

convert PRIntn int
convert PRUintn unsigned

convert PRSize size_t

convert PROffset32 int32_t
convert PROffset64 int64_t

convert PRPtrdiff ptrdiff_t

convert PRFloat64 double
2012-08-22 11:56:38 -04:00
Anant Narayanan
46105c0425 Bug 691234: Part 2/3: Implement WebRTC backend for MediaEngine on Desktop; r=jesup, r=roc 2012-07-12 04:53:08 -07:00
Anant Narayanan
23c187f947 Backout 038e89521330, bustage 2012-07-11 21:44:16 -07:00
Anant Narayanan
dcad8dd933 Bug 691234: Part 2/3: Implement WebRTC backend for MediaEngine on Desktop; r=jesup, r=roc 2012-07-11 21:22:24 -07:00