931b933075
Bug 1030324: Remove VP8 encoder resize work around. r=rjesup
...
The VP8 encoder was re-initialized completely when the frame size changed
instead of calling the vpx_enc_config_set() method. The work around is not
longer needed.
2015-09-08 11:02:40 -07:00
9704fc6569
Bug 1037997 - Added possibility to choose monitors during screen capturing. r=florian,jesup
...
--HG--
extra : rebase_source : 6b2476d5c03dffa27c8a5c570ec1392d332b81d2
2015-09-03 07:24:00 +02:00
c0fd8df08b
Bug 901633 - Part 11 - Add an API in webrtc.org's output mixer to get the output channel count. r=jesup
...
--HG--
extra : rebase_source : 15ca543bb0a7f3a0b79ab10531613cee605b3e7b
2015-09-01 14:25:48 +02:00
a84ad7251d
Bug 901633 - Part 9 - Make the necessary changes to VoEExternalMediaImpl::ExternalRecordingInsertData so that it the number of channels is forwarded down the webrtc.org code. r=jesup
...
--HG--
extra : rebase_source : 02a2e46f75db29f9333e1d36e462903e95d4c49b
2015-08-11 13:49:29 +02:00
834ece7e25
Bug 1198107 - Destroy VP8 encoder context before re-initing on resolution change to avoid leaking memory. r=jesup
...
--HG--
extra : transplant_source : Uv%0F7%E5N%C2%21%15b%5CF%12%E9%EE%B2%EE%D9%FB%D5
2015-08-25 16:55:28 +08:00
b865699f7f
Bug 1191872 - Move annotations to org.mozilla.gecko.annotation package, r=jchen
...
--HG--
rename : mobile/android/base/mozglue/JNITarget.java => mobile/android/base/annotation/JNITarget.java
rename : mobile/android/base/mozglue/RobocopTarget.java => mobile/android/base/annotation/RobocopTarget.java
rename : mobile/android/base/mozglue/WebRTCJNITarget.java => mobile/android/base/annotation/WebRTCJNITarget.java
2015-08-10 19:19:51 -04:00
f6d0d49348
Bug 1186657. r=jesup,nchen
2015-07-28 08:55:06 +02:00
3ea92ed1e2
Bug 1181265 - wallpaper over windows (driver?) returning null ptr to GetStreamCaps r=pkerr
2015-07-09 14:23:58 -04:00
b6bd5b4c75
Bug 881742: Fix null-deref on OOM (or bad allocation size due to corrupted video) in webrtc upstream code r=jesup
2015-06-18 00:06:36 -04:00
5d19fb925a
bug 1171143 - Fix iOS capture build. r=jesup
...
--HG--
extra : commitid : C2USxBeZe6F
extra : rebase_source : d217dcc4ab9dc6a9b36934e69f62291e22469b79
2015-02-28 19:47:07 -05:00
3fa2a3d032
Bug 1158372: clean up windows CreateCapabilityMap for video capture r=dmajor
2015-06-05 09:18:35 -04:00
e517cc3f1e
Bug 1166583 - Move chromium's MakeTuple function into namespace 'base' to avoid conflicts with mozilla::MakeTuple. r=froydnj
...
--HG--
extra : source : 2258a91d5781efe8e1d5f92f64ff173412705274
2015-05-09 21:09:40 -05:00
b8c6846e13
Bug 1162251: Fix WebRTC jitter buffer ignoring partial frames if the packet holds a complete NAL r=ehugg a=prep for uplift (kwierso)
2015-05-07 20:05:20 -04:00
770bb9731d
Bug 1158627 - WebRTC return error if GetEmptyFrame returns null r=jesup
2015-04-26 13:13:46 -07:00
4acbe13c9a
Bug 911450: sndio webrtc audio backend, build integration bits r=jesup
2015-04-10 21:36:54 +02:00
806c406d91
Bug 1143694 - Care about gralloc YV12 stride r=jesup
2015-04-02 09:28:11 -07:00
e6f385fb3d
Bug 1148527 - Indentation fix after bug 1145631, r=ehsan
2015-03-27 18:52:19 +00:00
76ac0daed4
Bug 1144912 - WebRTC Screenshare has own default FPS r=jib
2015-03-23 19:17:07 -07:00
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
e4a0d35e15
Bug 1136004 - Fix -Wthread-safety-analysis warning in webrtc. r=jesup
2015-03-02 19:51:29 -08:00
1482147cb9
Bug 1123012 - Just return a NULL ptr instead of casting NULL. r=jesup
2015-02-25 08:31:11 +01:00
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
4f3952c197
Bug 1130223 - Add an implementation to the big endian conditional. r=jesup
2015-02-15 09:36:00 +01:00
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
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
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
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
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
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
1c3409ae00
Bug 1122547 - Unbreak build on platforms missing std::llabs since bug 1089478. r=jesup
2015-01-16 17:41:00 -05:00
81ec239d97
Bug 1119852 - Don't forget to update _requestedCapability in Windows camera driver. r=jesup
2015-01-12 02:09:00 +01:00
Paul Kerr [:pkerr]
Pavlo
Paul Adenot
Andreas Pehrson
Mark Capella
Gian-Carlo Pascutto
Randell Jesup
Adrian Cruceru
Ted Mielczarek
Botond Ballo
Ethan Hugg
Landry Breuil
Sotaro Ikeda
Andrea Marchesini
Chris Peterson
Gian-Carlo Pascutto
Steve Singer
Randell Jesup
Jan Beich