896fc7df01
Bug 1141386 - Don't always assume base64 encoded EME key/Ids have padding stripped. r=edwin
2015-03-10 16:49:03 +13:00
f6c5da1b53
Bug 1140637: Add jsep_session_unittest to testing/cppunittest.ini, and unbust it. r=jesup
...
--HG--
extra : rebase_source : 91980628dd06114ab24be29c9b77629fef387619
extra : amend_source : 10cd425cafef72cc3555b27e1f05acf25e8ec856
2015-03-09 14:45:46 -07:00
0ac97b635c
Bug 1139132: Fix Chroma offset on WebRTC remote video when width is not even r=jesup
...
Fixed chroma plane offset calculation when frame width/height is not even
2015-03-03 21:06:00 +01:00
d35482f7d7
Bug 1140677 - Add RTPFragmentationHeader handling on gonk r=jesup
2015-03-09 18:36:23 -07:00
91323d7a02
Backed out changeset a1d51e3fea63 (bug 935838) for B2G test_udpsocket.html timeouts.
...
CLOSED TREE
--HG--
extra : rebase_source : c38820b067a8faf405bfae7f5b5fb1089bd29bbc
2015-03-09 16:35:06 -04:00
60fd94d4f2
Bug 1139779: Part1. Extract display dimension from SPS NAL. r=rillian
2015-03-10 21:19:41 +11:00
a2a4213345
Bug 935838 - Add per app network traffic statistics to the UDP socket. r=sicking, r=mayhemer
2015-03-06 06:38:00 -05:00
96a3f8c908
Bug 1140933 - Handle empty subsample encryption information in SampleIterator - r=cpearce
2015-03-09 13:55:33 +13:00
d2907f0381
Bug 1140797 - Prevent fatal assert when doing base64 decode in gmp-clearkey. r=edwin
2015-03-09 08:27:18 +13:00
50770662aa
Bug 1140797 - Make gmp-clearkey buildable outside of mozilla-central. r=edwin
2015-03-09 08:27:05 +13:00
13e1747bf3
Bug 1140648 - WebRTC check codec config max frame rate is set before using r=jesup
2015-03-06 19:05:11 -08:00
ca5799606e
Bug 1139144 - Remove unused empty() definition from databuffer.h. r=mt
...
--HG--
extra : rebase_source : 4ea88ac04ef457fbbe3707e9e7d9af511e107688
2015-03-03 12:49:37 -08:00
376726d034
Bug 1137932: Unwind the stack before starting the DTLS handshake. r=mt
...
--HG--
extra : rebase_source : c74e9a09e40c5a0ef9e00ca4dd326bc0ac8a4319
2015-02-27 15:17:23 -08:00
8eff7a3ee8
Bug 1135544 - Create an abstract base class for a track demuxer; r=kinetik
2015-03-05 17:30:44 +13:00
d19ee72543
Bug 1133051: Clean up SctpFlows on STS r=mt
...
--HG--
extra : rebase_source : cbc09ce944e4580e3e190766767612b98a8ced61
2015-02-13 13:32:01 -08:00
e8d1645b6c
Bug 1139087 - Add moz.build bugzilla metadata for codecs. r=kinetik,gps
2015-03-03 11:36:00 -08:00
63701128a7
Bug 1115483 - Accept a match on any a=fingerprint value. r=ekr
...
--HG--
extra : rebase_source : aff16495f2be12cb4c06777df39b39ad32fc4e2e
2015-01-12 15:53:59 -08:00
889621fe9a
Bug 1138777 - Don't do sync dispatch in gmp-clearkey AudioDecoder. r=edwin
2015-03-04 23:20:29 +13:00
ee400f62d6
Bug 1136107 - Handle reconfiguring audio device if it went away while the stream was stopped. r=padenot
2015-03-02 18:07:43 +13:00
fef42c5c7a
Bug 1137472: test vp9 sdp in sdp_unittests r=bwc
2015-03-03 23:46:16 -05:00
30c1cb6a48
Bug 1119973 - Update libvorbis to upstream 1.5.3. r=kinetik
2015-03-03 09:54:00 -08:00
00a74cff8f
Bug 1137472: Basic VP9 signaling/pipeline/conduit support r=bwc
2015-03-03 01:31:33 -05:00
c9b6a04ac4
Bug 1137474: Fix depacketization of "Generic" encoded RTP video r=pkerr
2015-03-03 01:31:33 -05:00
f91175ae7f
Bug 1137474: Basic vp9 support added to upstream (using 'generic' packetization) r=pkerr
2015-03-03 01:31:33 -05:00
9b19093204
Bug 1129263 - Part 6. Remove DOMMediaStream::TrackTypeHints. r=roc,jesup
2015-02-09 15:23:34 +08:00
68db4947b3
Bug 1129263 - Part 5. Add intial remote PeerConnection tracks atomically to SourceMediaStream. r=jesup
2015-02-11 16:21:11 +08:00
e4a0d35e15
Bug 1136004 - Fix -Wthread-safety-analysis warning in webrtc. r=jesup
2015-03-02 19:51:29 -08:00
c1cdd99c64
Bug 1133866: Some refactoring and simplification in JsepSessionImpl. r=mt
...
--HG--
extra : rebase_source : 64e445b182c7b1fad514d354cc6bf1f4abfecd7f
2015-02-25 08:36:01 -08:00
16d71fc156
Backed out changeset a622dbe33efb (bug 1135544) for ASAN mochitest-3 bustage on CLOSED TREE
2015-03-02 18:13:39 +05:30
ad3b3d35da
Bug 1135544 - Create an abstract base class for a track demuxer; r=kinetik
2015-03-02 16:34:44 +13:00
1618c6fa37
Bug 1137489 - Fix unified build failure in gmp-clearkey. r=edwin
2015-02-28 16:15:18 +13:00
e4ef49ff55
Bug 1136986 - Fix unthreadsafe uses of GMPVideoHost in gmp-clearkey. r=edwin
2015-02-28 10:23:33 +13:00
b7a81f04b3
Bug 1137508 Change H264 maxPayloadSize to 0 for Mode 1 r=jesup
2015-02-26 15:29:36 -08:00
573c0c5d46
Bug 1130175 - nICEr: avoid sysctl.h include. r=bwc
2015-02-05 22:24:20 +01:00
1482147cb9
Bug 1123012 - Just return a NULL ptr instead of casting NULL. r=jesup
2015-02-25 08:31:11 +01:00
66edb0fc3b
Bug 1135942 - Enable MMX/SSE code unconditionally in libsoundtouch, it does runtime detection anyways. r=padenot
2015-02-25 13:12:15 +09:00
1c767dd0c0
Bug 1135902: Set stream id on fake media streams. r=drno
...
--HG--
extra : rebase_source : edf3e19236c05d6040ae2320d7f385105a30efe2
extra : amend_source : 79efc1ed9537b5710516e9883dc91f45c344847e
2015-02-23 15:19:17 -08:00
5bff0c0b03
Bug 1135753 - Mark some overridden virtual functions in WebRTC as MOZ_OVERRIDE; r=mt
2015-02-24 09:51:57 -05:00
bbfbfbf28c
Bug 1135878 - Simply post-error cleanup logic in WASAPI cubeb backend. r=padenot
2015-02-24 13:42:33 +13:00
05c71da4ba
Bug 1136004: fix TSAN warning in webrtc when RED isn't enabled r=cpeterson
2015-02-24 02:08:04 -05:00
71cad0a045
Bug 1134991 - Failure to set up voice communication mode in OpenSLES should not be fatal. r=jesup
2015-02-20 19:13:13 +01:00
1d16a313f3
Bug 1128116: Fix decoding H264 in webrtc where SPS & PPS aren't in a STAP-A packet r=ehugg
...
FF 37 and before didn't encode SPS/PPS into a STAP-A packet, and the
webrtc.org in branch 40 code doesn't handle that (common) case.
2015-02-22 19:10:59 -05:00
911ef245b5
Bug 1061339 - Part 2: Build AVX code on all Windows build. r=rillian, r=ted
2015-02-16 23:10:00 -05:00
fcc34bf07a
Bug 1061339 - Part 1: Always use VS2013 target. r=rillian
...
--HG--
rename : media/libvpx/vp8_rtcd_x86-win32-vs8.h => media/libvpx/vp8_rtcd_x86-win32-vs12.h
rename : media/libvpx/vp8_rtcd_x86_64-win64-vs8.h => media/libvpx/vp8_rtcd_x86_64-win64-vs12.h
rename : media/libvpx/vp9_rtcd_x86-win32-vs8.h => media/libvpx/vp9_rtcd_x86-win32-vs12.h
rename : media/libvpx/vp9_rtcd_x86_64-win64-vs8.h => media/libvpx/vp9_rtcd_x86_64-win64-vs12.h
rename : media/libvpx/vpx_config_x86-win32-vs8.asm => media/libvpx/vpx_config_x86-win32-vs12.asm
rename : media/libvpx/vpx_config_x86-win32-vs8.h => media/libvpx/vpx_config_x86-win32-vs12.h
rename : media/libvpx/vpx_config_x86_64-win64-vs8.asm => media/libvpx/vpx_config_x86_64-win64-vs12.asm
rename : media/libvpx/vpx_config_x86_64-win64-vs8.h => media/libvpx/vpx_config_x86_64-win64-vs12.h
rename : media/libvpx/vpx_scale_rtcd_x86-win32-vs8.h => media/libvpx/vpx_scale_rtcd_x86-win32-vs12.h
rename : media/libvpx/vpx_scale_rtcd_x86_64-win64-vs8.h => media/libvpx/vpx_scale_rtcd_x86_64-win64-vs12.h
2015-02-16 21:15:00 -05:00
f11edd5123
Bug 1132813 - Enabling DTLS 1.2 for WebRTC, r=abr
2015-02-21 10:35:21 +13:00
b25a2bb677
Bug 1089798 - MediaStream ID tests. r=bwc
2015-02-17 22:54:00 -05:00
629772a073
Bug 1089798 - Implemenation of MediaStream IDs. r=bwc
2015-02-19 12:59:00 -05:00
2880c3a9b1
Bug 1124031 part 4 - Enforce min CDM version from keySystem string. r=bz
2015-02-20 14:38:08 +13:00
fc5089207f
Bug 1134064: Part3. Don't evict partial data and make resource unplayable. r=k17e
2015-02-20 14:19:14 +13:00
0bdf32b0cd
Bug 1133386 - Introduce an XASSERT() macro to libcubeb rather than (ab)using assert(). r=padenot f=dmajor
2015-02-20 13:42:14 +13:00
dc0853c72b
Backed out changeset abf7a473323c (bug 1089798)
...
--HG--
extra : rebase_source : c1248ca0d1b5f136c935a606f2968cc16aed2b7d
2015-02-19 10:31:42 +01:00
51d447f32c
Backed out changeset a6ccfddbdac0 (bug 1089798)
...
--HG--
extra : rebase_source : bec618ddce6083d609ae1432b2b7c7366634a508
2015-02-19 10:31:37 +01:00
a6a9beffd3
Bug 1124411 - Add timeout when calling WaitForMultipleObjects to wallpaper over a deadlock. r=kinetik
2015-02-19 19:35:07 +13:00
eda5e9c69f
Bug 1134102 - Move cubeb's stream reconfiguration to render thread. r=padenot
2015-02-18 18:43:07 +13:00
67200dd25c
Bug 1133190 - Don't use auto_unlock in paths where lock is being destroyed. r=kinetik
2015-02-19 19:35:06 +13:00
c3491769e4
Bug 1132034 - Properly round the number of frame for the accumulating clock in WASAPI. r=kinetik
2015-02-19 19:35:06 +13:00
f7c07beb8f
Bug 1134078 - Don't restart stopped cubeb streams when handling device change notifications. r=padenot
2015-02-18 16:06:55 +13:00
daec70e513
Bug 1089798 - MediaStream ID tests. r=bwc
2015-02-17 22:54:00 -05:00
0084d83191
Bug 1089798 - Implemenation of MediaStream IDs. r=bwc
2015-02-17 22:52:00 -05:00
aa31157b0a
Bug 1130450: Properly handle MP4 with Apple QT extension. r=k17e
2015-02-19 15:37:11 +13:00
4f3952c197
Bug 1130223 - Add an implementation to the big endian conditional. r=jesup
2015-02-15 09:36:00 +01:00
28c7d528c1
Bug 1133572 - Use new demuxer for all sample fetches; r=cpearce
2015-02-18 19:13:15 +13:00
250218269a
Bug 1133572 - Remove duplication of logic from GetNextKeyframeTime(); r=cpearce
2015-02-18 19:13:14 +13:00
8b8ae2538d
Bug 1132257 - Update cubeb from upstream. r=padenot
2015-02-17 11:52:59 +13:00
794f966f80
Bug 1130534: Use a single bidirectional media conduit that MediaPipelines can attach/detach at will. r=jesup
...
--HG--
extra : rebase_source : 202a83e513d88bc14f1be2c5b438998461ff4a50
2015-02-10 10:11:24 -08:00
248d6e6d79
Bug 1017888 - Part 2: Testing for renegotiation. r=mt, r=drno
...
--HG--
extra : rebase_source : 7434ef35ea6294966220f20431941e0790c4221c
2015-02-10 10:17:03 -08:00
1f815978b4
Bug 1017888 - Part 1: Renegotiation support. r=mt, r=smaug
...
--HG--
extra : rebase_source : df1c2962ee88f75c6ad676b9cd79978a87dafb65
extra : amend_source : c938904331323ff3565624795ac76d82502f43fb
2014-12-10 15:53:54 -08:00
1de69a46f1
Bug 1133479 - Fixed media/gmp-clearkey build on mingw. r=cpearce
2015-02-17 11:18:04 +01:00
6dffa7b1fa
Bug 1133478: Postpone parsing TRUN atom until we have all dependent atoms. r=k17e
...
Atoms may be out of order inside MOOF, in particular the TFDT may only be
defined after TRUN.
2015-02-17 16:22:52 +13:00
7eb58b57db
Bug 1133291 - Remove unused code from Clear Key's openaes. r=cpearce
2015-02-15 22:07:10 -08:00
2cd7434422
Bug 1127554 - Do a fallible allocation in SampleIterator::GetNext. v1 r=mattwoodrow
2015-02-13 12:19:38 -08:00
ced4bbc434
Bug 1127554 - Make MP4Sample::Replace fallible. v1 r=mattwoodrow
2015-02-13 12:19:37 -08:00
6217aabf6a
Bug 1127554 - Make MP4Sample::Prepend fallible. v1 r=mattwoodrow
2015-02-13 12:19:37 -08:00
620ec20063
Bug 1127554 - Make MP4Sample::Pad fallible. v1 r=mattwoodrow
2015-02-13 12:19:36 -08:00
f055cc749a
Bug 1127554 - Get rid of infallible allocation in MP4Sample copy constructor. v2 r=mattwoodrow
2015-02-13 12:19:36 -08:00
cc4a1f03e4
Bug 1131960 - Check for NEON capability before using NEON code. r=derf
...
CLOSED TREE
2015-02-13 05:13:00 -05:00
40f518f5bf
Bug 1108248: Swap CreateTimerQueueTimer() for timerSetEvent() in webrtc win32 code r=dmajor
...
Avoids limits on the number of realtime (timerSetEvent()) timers
2015-02-06 17:24:50 -05:00
e29d8e1845
Bug 1127642 WebRTC support for H.264 max_mbps r=jesup
2015-02-12 11:14:57 +08:00
687807441c
Bug 1132193: Re-enable AEC debug logging in getUserMedia r=pkerr
...
Temporarily disabled by landing for upstream webrtc branch 40. Also saves
as .wav format now
2015-02-12 07:46:59 -05:00
c2512b6c85
Bug 1128939: Part3. Allocate media buffer internal memory dynamically. r=k17e
2015-02-12 18:52:12 +11:00
4132c3064d
Bug 1128939: Part2. Make sure we limit read to buffer size and handle error nicely. r=k17e
2015-02-12 18:52:12 +11:00
670653d955
Bug 1131788 - cubeb: Unable to use InterlockedAdd64 on MSVC2010 (and mingw). r=padenot
2015-02-10 17:45:00 +13:00
c7a5446fd2
Bug 1124175: Remove limits on odd webrtc downsample sizes due to load/bitrate r=pkerr
...
Also convert assert to limits on max size
2015-02-11 17:29:01 -05:00
5e60171da8
Bug 1131871 - auto_com should only uninitialize when successful. r=padenot
...
--HG--
extra : rebase_source : 4af27a3e50b68ad1ee4d869b1a3bee3eeb4a1141
2015-02-12 10:33:45 +13:00
11407de1f6
Bug 1130917 - Part 2 - improve error handling of StoreData() and ReadData(). r=edwin.
2015-02-10 18:18:00 +01:00
dbc2bf7237
Bug 1128939: Part1. Ensure we have any space in the media buffer before writing. r=k17e
2015-02-11 17:40:13 +11:00
c32cf025e7
Bug 1131768 - Unlock before tearing down the stream in case of error, to avoid recursive locking. r=kinetik
2015-02-11 00:12:09 +01:00
9ec922dcb3
Bug 1129791 - Check connection state rather than context state. r=ekr
...
--HG--
extra : transplant_source : %E5RDV%AF%3B%9D%7C%0F%10%9BF3%BB%29%06%8C%92%CF1
2015-02-05 17:18:57 +11:00
554ea67ff5
Bug 1131340 - Avoid template aliasing since GCC 4.6 lacks support. r=cpearce
2015-02-10 14:27:36 +13:00
242708cf72
Bug 1127201 (attempt 2, part 1) - Replace most NS_ABORT_IF_FALSE calls with MOZ_ASSERT. r=Waldo.
...
--HG--
extra : rebase_source : 488e401ff87e31a2074c4108c4df0572d9536667
2015-02-09 14:34:50 -08:00
5098817b4c
Bug 1127213 - Fix various issues with the device change notification in the WASAPI cubeb backend. r=kinetik
...
This patch does the following:
- Introduces an owned_critical_section object to be able to assert that a
current thread owns a critical section
- Change the auto_lock to use the above, add auto_unlock (basically like the
Gecko AutoEnter/AutoExit things)
- Fix an issue during audio output device switch where the clock would use the
old sample rate. Apparently I did not notice this because my headset and the
sound card on this laptop have the same rate
- Check that we could acquire a device_enumerator in the ctor before
deallocating in the dtor, as that can happen if a ton of streams are running at
once (I had this issue running the full mochitest suite)
- Stop getting another device_enumator in unregister_notification_client, fixing a leak
- Assert that setup_wasapi_stream and close_wasapi_stream are called with the lock held, this was the cause of the crash for this bug
- Make close_wasapi_stream clear out its state to make sure setup_wasapi_stream
and close_wasapi_stream are called in the right order (especially, not two
setup_wasapi_stream without close in between, since that would leak stuff)
- In wasapi_stream_destroy, unregister the notification client before destroying
the CRITICAL_SECTION (this was the cause of a crash I duped against this bug)
2015-02-09 14:42:43 +01:00
b514443060
Backed out changeset 2f46afa97421 (bug 1127213) for another bustage on a CLOSED TREE
2015-02-09 16:51:14 +01:00
14201f3594
Bug 1123768 - Backout bug 1108455 to avoid truncating the end of audio streams on Vista+. r=kinetik
2015-02-09 14:43:03 +01:00
8b005f3fb1
Bug 1127213 - Fix various issues with the device change notification in the WASAPI cubeb backend. r=kinetik
...
This patch does the following:
- Introduces an owned_critical_section object to be able to assert that a
current thread owns a critical section
- Change the auto_lock to use the above, add auto_unlock (basically like the
Gecko AutoEnter/AutoExit things)
- Fix an issue during audio output device switch where the clock would use the
old sample rate. Apparently I did not notice this because my headset and the
sound card on this laptop have the same rate
- Check that we could acquire a device_enumerator in the ctor before
deallocating in the dtor, as that can happen if a ton of streams are running at
once (I had this issue running the full mochitest suite)
- Stop getting another device_enumator in unregister_notification_client, fixing a leak
- Assert that setup_wasapi_stream and close_wasapi_stream are called with the lock held, this was the cause of the crash for this bug
- Make close_wasapi_stream clear out its state to make sure setup_wasapi_stream
and close_wasapi_stream are called in the right order (especially, not two
setup_wasapi_stream without close in between, since that would leak stuff)
- In wasapi_stream_destroy, unregister the notification client before destroying
the CRITICAL_SECTION (this was the cause of a crash I duped against this bug)
2015-02-09 14:42:43 +01:00
67411f1d5e
Bug 1129732: Part1. Dynamically adjust calculations using timestampoffset. r=mattwoodrow
...
Timestamp Offset calculations are now done exclusively by the Media Source
components which allow to recalculate them on the fly. By abstracting those
offsets it remove the need for the sub-decoders to handle them (which allows
to add WebM support).
2015-02-09 23:28:59 +11:00
2b8def7fd2
Bug 1129071: Add log error should we run out of memory. r=k17e
2015-02-09 23:28:58 +11:00
a85bba5efb
Bug 1130290 - Remove PeerConnectionImpl::CreateFakeMediaStream. r=jesup
...
--HG--
extra : rebase_source : c5fe9a894178e600c48ce22e45b9c124c76cf712
2015-02-05 23:56:00 +01:00
d3826daa16
Back out Bug 1127201 (part 2) for various problems.
2015-02-06 15:04:32 -08:00
d34f0301b8
Bug 1127201 (part 2) - Convert all NS_ABORT_IF_FALSE calls to MOZ_ASSERT. r=Waldo.
...
--HG--
extra : rebase_source : 99182e70335d2b5ff95f8c528ae992d37294be3a
2015-02-04 20:05:36 -08:00
9134477c28
Bug 1129921 - Account for stopCapture possibly being called twice. r=jesup
2015-02-05 18:24:02 +01:00
58eb5e24e0
Bug 1129858 - Get the local preview surface (line dropped during merge). r=jesup
2015-02-05 18:24:02 +01:00
e8ec6fb3c4
Bug 1129365 - Don't assume setPictureSize supports the same sizes as setPreviewSize. r=jesup
2015-02-05 18:24:02 +01:00
847dbb9825
Bug 1120796 - Part 1: Prepare code for explicit bool operators. r=Waldo
2015-02-03 18:52:28 +02:00
82f6804a0c
Bug 1075199 - Output a different clearkey.info depending on platform - r=cpearce,gps
...
--HG--
rename : media/gmp-clearkey/0.1/clearkey.info => media/gmp-clearkey/0.1/clearkey.info.in
2015-02-03 16:59:39 +13:00
bc2814dfe3
Bug 1075199 - More logging in ClearKey CDM - r=cpearce
2015-01-27 18:35:36 +13:00
4e3006ddd2
Bug 1075199 - WMF decoding in ClearKey CDM - r=cpearce
2015-01-16 10:37:54 +13:00
f68bdd6433
Bug 1075199 - Import WMF decoding code from cpearce's gmp-clearkey implementation - r=cpearce
2015-01-16 10:37:54 +13:00
029454fbd8
Bug 1128410: Part2. Make memory allocation dynamic. r=kentuckyfriedtakahe
...
Allocations are fallibles.
2015-02-02 21:36:51 +11:00
bc3b53cf6a
Bug 1128410: Lazily allocate the MP4 parser buffer. r=kentuckyfriedtakahe
...
This buffer is unsused for fragmented MP4. So we don't need to unecessarily
allocate it and block a chunk of 3MB data. Also, this removes the restriction
of playing YouTube video > 1080p
2015-02-02 21:36:48 +11:00
96d001f3d4
Bug 1120065 - unit test for TURN deallocation. r=bwc
2015-01-30 12:36:00 +01:00
d6b2b87209
Bug 1120065 - release TURN resources on PC release. r=bwc
2015-01-27 01:18:00 +01:00
ba151bbff5
Bug 1120065 - removed dead function nr_ice_candidate_destroy_cb. r=bwc
2015-01-23 13:47:00 +01:00
a35dbaeebf
Bug 1126593 - Add a global fallible instance, so that using fallible works directly, everywhere. r=njn
...
--HG--
rename : memory/mozalloc/fallible.h => memory/fallible/fallible.h
2015-02-02 09:56:13 +09:00
4d6309bd93
Bug 1083658 - add "output-downscaled" to GMP. r=cpearce.
2015-02-01 09:18:39 +08:00
c1dd9140a5
Bug 1121332. Part 1 - add media key status to gmp-api. r=cpearce.
2015-01-31 13:22:12 +13:00
fcfce237f4
Bug 1127115 - Make MP4 parser assertion non-fatal - r=ajones
2015-01-30 16:54:12 +13:00
0dd312ac8c
Bug 1099318: Fix for conduit receive then send creation order issue. Now insensitive to order. r=bwc"
2015-01-29 08:52:40 -08:00
adfc170313
Bug 1109248: Merge with webrtc.org update (android compile/merge fixes) r=jesup
...
ON A CLOSED TREE
2015-01-29 18:34:16 -05:00
f82b381b47
Bug 1109248: remove unused media/webrtc/trunk/base directory (ancient) rs=jesup
2015-01-29 18:33:36 -05:00
96d17ba850
Bug 1109248: Include/etc fixes for B2G from webrtc.org update rs=jesup
2015-01-29 18:33:36 -05:00
9494dbe58e
Bug 1109248: webrtc.org update needs some floating point routines on Android rs=jesup
2015-01-29 18:33:36 -05:00
a333fa4da0
Bug 1109248: Merge webrtc.org update with our OpenSLES changes rs=jesup
2015-01-29 18:33:36 -05:00
2bd1c1b6a1
Bug 1109248: fixes for changes to webrtc Android camera fps handling r=jesup
2015-01-29 18:33:36 -05:00
715dfa95f8
Bug 1109248: Revert removal of SetAndroidObjects calls in webrtc.org r=jesup
2015-01-29 18:33:36 -05:00
37194f082b
Bug 1109248: Adapt GMP video decoder code to API changes in webrtc.org 40 r=ehugg
2015-01-29 18:33:36 -05:00
d17d6d6c85
Bug 1109248: basic adapation of new webrtc/base directory to build in mozilla rs=jesup
2015-01-29 18:33:36 -05:00
5fa8bc8fb9
Bug 1109248 - build fixes for OpenBSD r=jesup
...
- check for __GLIBC__ instead of __GLIBCXX__ to include <execinfo.h>
- check for WEBRTC_BSD instead of BSD to include <stdlib.h>
2015-01-29 18:33:36 -05:00
47e542881b
Bug 1109248: basic compile fixes for webrtc.org 40 update rs=jesup
...
Mostly #ifdefing Chrome-specific code and replacing WEBRTC_TRACE with LOG_F/etc
2015-01-29 18:33:36 -05:00
ead017e967
Bug 1109248: gyp changes to adapt to webrtc.org 40 update r=ted
2015-01-29 18:33:36 -05:00
100c8393ed
Bug 1109248: revert removal of webrtc audio ExternalRecording interface rs=jesup
2015-01-29 18:33:36 -05:00
7fa6134aa7
Bug 1109248: Revert webrtc upstream Issue 18399004 which removed APIs we're using rs=jesup
...
https://webrtc-codereview.appspot.com/18399004
2015-01-29 18:33:36 -05:00
baec6cfbd0
Bug 1109248: Rollup of changes previously applied to media/webrtc/trunk/webrtc rs=jesup
2015-01-29 18:33:36 -05:00
a50873f485
Bug 1109248: Webrtc updated to branch 40 7864; pull made Wed Dec 10 12:23:33 EST 2014 rs=jesup
...
--HG--
rename : media/webrtc/trunk/webrtc/system_wrappers/interface/thread_annotations.h => media/webrtc/trunk/webrtc/base/thread_annotations.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestFEC.h => media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestRedFec.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/accelerate.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/accelerate.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/accelerate.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/accelerate.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/audio_decoder.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/audio_decoder_impl.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/audio_decoder_impl.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/audio_decoder_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/audio_multi_vector.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_multi_vector.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/audio_multi_vector.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_multi_vector.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/audio_multi_vector_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/audio_vector.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_vector.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/audio_vector.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_vector.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/audio_vector_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_vector_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/background_noise.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/background_noise.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/background_noise.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/background_noise.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/background_noise_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/background_noise_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/buffer_level_filter.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/buffer_level_filter.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/buffer_level_filter.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/buffer_level_filter_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/buffer_level_filter_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/comfort_noise.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/comfort_noise.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/comfort_noise.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/comfort_noise.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/comfort_noise_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/comfort_noise_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/decision_logic.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/decision_logic.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/decision_logic_fax.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_fax.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/decision_logic_fax.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_fax.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/decision_logic_normal.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/decision_logic_normal.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_normal.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/decision_logic_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/decoder_database.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decoder_database.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/decoder_database.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decoder_database.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/decoder_database_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/defines.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/defines.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/delay_manager.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_manager.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/delay_manager.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_manager.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/delay_manager_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/delay_peak_detector.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/delay_peak_detector.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_peak_detector.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/delay_peak_detector_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_peak_detector_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/dsp_helper.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dsp_helper.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/dsp_helper.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dsp_helper.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/dsp_helper_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dsp_helper_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/dtmf_buffer.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_buffer.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/dtmf_buffer_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/expand.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/expand.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/expand.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/expand.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/expand_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/expand_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/interface/audio_decoder.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/interface/neteq.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/interface/neteq.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/merge.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/merge.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/merge.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/merge.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/merge_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/merge_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/mock/mock_audio_decoder.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/mock/mock_audio_vector.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_audio_vector.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/mock/mock_buffer_level_filter.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/mock/mock_decoder_database.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/mock/mock_delay_manager.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/mock/mock_delay_peak_detector.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/mock/mock_dtmf_buffer.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/mock/mock_dtmf_tone_generator.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/mock/mock_external_decoder_pcm16b.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/mock/mock_packet_buffer.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/mock/mock_payload_splitter.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_payload_splitter.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/neteq_impl.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_impl.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/neteq_stereo_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/neteq_tests.gypi => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_tests.gypi
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/normal.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/normal.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/normal.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/normal.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/packet.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/packet_buffer.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet_buffer.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/packet_buffer_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/payload_splitter.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/payload_splitter.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/payload_splitter.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/payload_splitter.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/payload_splitter_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/post_decode_vad.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/post_decode_vad.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/post_decode_vad.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/post_decode_vad.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/post_decode_vad_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/post_decode_vad_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/preemptive_expand.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/preemptive_expand.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/preemptive_expand.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/preemptive_expand.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/random_vector.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/random_vector.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/random_vector.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/random_vector.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/random_vector_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/random_vector_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/rtcp.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/rtcp.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/statistics_calculator.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/statistics_calculator.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/statistics_calculator.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/sync_buffer.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/sync_buffer.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/sync_buffer.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/sync_buffer.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/sync_buffer_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/sync_buffer_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/test/neteq_performance_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/time_stretch.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/time_stretch.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/time_stretch.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/time_stretch.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/time_stretch_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/time_stretch_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/timestamp_scaler.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/timestamp_scaler.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/timestamp_scaler.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/timestamp_scaler_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/tools/audio_loop.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/tools/audio_loop.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/tools/input_audio_file.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/input_audio_file.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/tools/input_audio_file.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/input_audio_file.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/tools/neteq_performance_test.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/tools/neteq_performance_test.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/tools/neteq_rtpplay.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/tools/rtp_generator.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_generator.cc
rename : media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_ios.cc => media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_ios.mm
rename : media/webrtc/trunk/webrtc/modules/video_capture/ios/video_capture_ios_objc.h => media/webrtc/trunk/webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.h
rename : media/webrtc/trunk/webrtc/modules/video_capture/ios/video_capture_ios_objc.mm => media/webrtc/trunk/webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.mm
rename : media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h => media/webrtc/trunk/webrtc/system_wrappers/interface/rtp_to_ntp.h
rename : media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp.cc => media/webrtc/trunk/webrtc/system_wrappers/source/rtp_to_ntp.cc
rename : media/webrtc/trunk/webrtc/test/mac/run_tests.mm => media/webrtc/trunk/webrtc/test/mac/run_test.mm
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq4/post_decode_vad_unittest.cc => media/webrtc/trunk/webrtc/test/run_test.cc
2015-01-29 18:33:35 -05:00
14e52a60de
Bug 1095218 - Part 2: Multistream support. r=mt
...
--HG--
extra : rebase_source : d699a4408c351014e30be3b3dfe148bda14c643f
2014-12-10 11:17:09 -08:00
3a5fa56125
Bug 1095218 - Part 1: msid support. r=mt
...
--HG--
extra : rebase_source : 7b5cd5efdaec5d53dd4d39aa1f4226659c06f1cb
2014-12-01 21:19:57 -08:00
f37df5625b
Backed out changeset 774ff21aecb3 (bug 1120128) for build bustage
2015-01-27 17:48:43 -08:00
e0f2ca34db
Bug 1120128 - Implement blacklist for DXVA and blacklist AMD Radeon HD 5800. r=Bas
...
--HG--
extra : rebase_source : 782dc73a9f284c7a0d98061f2fe2dfb87e3bb149
2015-01-26 13:34:28 +13:00
98910d890f
Bug 1099414: Ensure that NrSocketIpc is destroyed on STS, for consistency. r=ekr
...
--HG--
extra : rebase_source : e706370706dd0d9ec938798966c752f298a260bf
2014-12-23 16:22:02 -08:00
c66aed798c
Bug 1099414: Use RefPtr logic instead of delete when nr_socket_local_create fails. r=ekr
...
--HG--
extra : rebase_source : 8be15d27527166b0dfb322b90e6f5244bcabcc95
2014-11-14 15:58:56 -08:00
971f76a2b4
Bug 1126036: Queue runnables for starting gathering and checking in PCMedia until the proxy lookup is complete. r=mt
...
--HG--
extra : rebase_source : 3265e13f669d08c663ab908cf96d3fb26a683f16
2015-01-26 15:24:37 -08:00
bfc4d164a1
Bug 1124939 - Add "individualization-request" to MediaKeyMessageType. r=bz
2015-01-26 20:08:00 +01:00
e49b1630b3
Bug 1118383 - Plug memory leak in openaes - r=cpearce
2015-01-27 19:10:11 +13:00
629afc55c1
Bug 1118597 - Parse sinf boxes in MoofParser - r=jya
2015-01-19 21:39:00 +13:00
d7eb4c6718
Bug 1118597 - Re-enable MoofParser for encrypted MP4 streams - r=jya
2015-01-27 18:35:36 +13:00
1085e3e63f
Bug 1125047 - GMP should catch decoder failures r=jesup
2015-01-26 15:00:06 -08:00
bf8ba31d3f
Bug 949703 - Part 2: Consolidate the two copies of DummySocket we have floating around. r=drno
...
--HG--
extra : rebase_source : 53ff83c3f788dfb06e5fee0a276176f8bac805fd
2014-12-19 11:11:02 -08:00
f649bed22d
Bug 949703 - Part 1: Use HTTP proxy for WebRTC TURN/TCP r=ekr
...
--HG--
extra : rebase_source : 607cd8e262fad408e605114ed0dabad249a10ec9
2015-01-21 16:26:00 -08:00
da64cbe550
Bug 1124542 - WebrtcGmpVideoDecoder shouldn't crash when GMP completion callbacks are received. r=rjesup
2015-01-21 20:26:00 +13:00
f29aff15e4
Bug 1124021 - Fix dangerous UniquePtr usage pattern in GMP. r=cpearce
2015-01-20 18:39:00 +13:00
a1d6e2bdbb
Bug 1124518 - Update libnestegg. r=ehsan
2015-01-22 15:04:39 +13:00
Chris Pearce
Byron Campen [:bwc]
Cesar Guirao
Sotaro Ikeda
Ryan VanderMeulen
Jean-Yves Avenard
Dragana Damjanovic
Edwin Flores
Ethan Hugg
EKR
Anthony Jones
Ralph Giles
Martin Thomson
Matthew Gregan
Randell Jesup
Andreas Pehrson
Chris Peterson
Nigel Babu
Felix Janda
Gian-Carlo Pascutto
Mike Hommey
Ehsan Akhgari
Makoto Kato
Nils Ohlmeier [:drno]
Carsten "Tomcat" Book
Paul Adenot
Steve Singer
Jacek Caban
Bobby Holley
Gian-Carlo Pascutto
Randell Jesup
Karina Li
Matthew Gregan
David Major
JW Wang
Nicholas Nethercote
Andrew McCreight
Birunthan Mohanathas
Edwin Flores
Paul Kerr [:pkerr]
Landry Breuil
Wes Kocher
Matt Woodrow
