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  • a15fbfdcde Add round-robin selection of send stream to pad on. stefan@webrtc.org 2014-06-17 17:32:05 +00:00
  • 9c09e6ee2b Add high perf mode to VP8 niklas.enbom@webrtc.org 2014-06-17 16:32:08 +00:00
  • 26eaf7c7f7 Add a check to all.gyp to respect the include_tests variable. andrew@webrtc.org 2014-06-17 16:10:20 +00:00
  • 2eaac188bb Makes the sid of a closed DataChannel available to reuse per the spec. jiayl@webrtc.org 2014-06-17 16:02:46 +00:00
  • a685c9df62 base: Renaming + conforming: post commit review changes for https://webrtc-codereview.appspot.com/17699005/ henrike@webrtc.org 2014-06-17 14:48:44 +00:00
  • 5654b305e5 Rebase webrtc/base with r6464 version of talk/base: cd webrtc/base svn diff -r 6463:6464 http://webrtc.googlecode.com/svn/trunk/talk/base > 6464.diff patch -p0 -i 6464.diff henrike@webrtc.org 2014-06-17 14:37:05 +00:00
  • d469443959 Rolling new version of opus.gyp tina.legrand@webrtc.org 2014-06-17 13:24:48 +00:00
  • ed3e0d8f0d Increasing tolerances quite a bit to fight flakes. phoglund@webrtc.org 2014-06-17 11:09:00 +00:00
  • ae740dd94c (Auto)update libjingle 69359922-> 69365993 buildbot@webrtc.org 2014-06-17 10:56:41 +00:00
  • d42da54768 Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..." minyue@webrtc.org 2014-06-17 09:50:12 +00:00
  • 851a09e71a Initial GN work for WebRTC kjellander@webrtc.org 2014-06-17 08:54:03 +00:00
  • 2ca2188906 Restore ptypes.txt file henrik.lundin@webrtc.org 2014-06-17 08:51:01 +00:00
  • 6b061425c2 Updated W3C getusermedia tests to the latest version of the spec. phoglund@webrtc.org 2014-06-17 08:46:58 +00:00
  • 8f8503d947 Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling. minyue@webrtc.org 2014-06-17 08:02:05 +00:00
  • 44a317a698 (Auto)update libjingle 69337301-> 69359922 buildbot@webrtc.org 2014-06-17 07:49:15 +00:00
  • 9f36c087f1 Makes it possible to prevent some third party libraries (jsoncpp and openssl) from being linked. This makes it possible to link webrtc with external implementations of those libraries in case the project depending on webrtc requires another version of those libraries. henrike@webrtc.org 2014-06-16 21:35:20 +00:00
  • 53f57936c1 (Auto)update libjingle 69306183-> 69323802 buildbot@webrtc.org 2014-06-16 21:08:51 +00:00
  • 587ef60056 Implement RTP extension support in WebRtcVideoEngine2. pbos@webrtc.org 2014-06-16 17:32:02 +00:00
  • d054bff3b9 (Auto)update libjingle 69292418-> 69293749 buildbot@webrtc.org 2014-06-16 14:37:41 +00:00
  • d980307197 Add max limit of number for overuses. When limit is reached always apply the rampup delay. asapersson@webrtc.org 2014-06-16 14:27:19 +00:00
  • 88d9fa63df (Auto)update libjingle 69291002-> 69292418 buildbot@webrtc.org 2014-06-16 14:11:32 +00:00
  • 4b12d40008 Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class. asapersson@webrtc.org 2014-06-16 14:09:28 +00:00
  • 27626a6256 (Auto)update libjingle 69278008-> 69291002 buildbot@webrtc.org 2014-06-16 13:39:40 +00:00
  • d6e2213edd Remove ivinnichenko from webrtc/test/OWNERS kjellander@webrtc.org 2014-06-16 11:42:27 +00:00
  • 1e3c5c248a Importing ThreadChecker class from Chromium henrik.lundin@webrtc.org 2014-06-16 11:34:44 +00:00
  • b099a6f9ab Adds aluebs@webrtc.org as owner to audio_processing bjornv@webrtc.org 2014-06-16 10:30:58 +00:00
  • 721f970cba common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16 bjornv@webrtc.org 2014-06-16 10:30:14 +00:00
  • eb16b811fb Implements start bitrate for new video API. mflodman@webrtc.org 2014-06-16 08:57:39 +00:00
  • 0a1e7e0b00 (Auto)update libjingle 69276003-> 69278008 buildbot@webrtc.org 2014-06-16 08:34:09 +00:00
  • 63e46077a3 Add thread annotations to parts of ACMGenericCodec henrik.lundin@webrtc.org 2014-06-16 08:02:25 +00:00
  • 249211eec6 Disable flaky test (WebRtcVideoMediaChannelTest.GetStats) on DrMemory Full. asapersson@webrtc.org 2014-06-16 07:56:17 +00:00
  • d159140965 (Auto)update libjingle 69260070-> 69276003 buildbot@webrtc.org 2014-06-16 07:49:00 +00:00
  • 2bae3211b1 Add missing sources to webrtc/base/base.gyp kjellander@webrtc.org 2014-06-16 07:11:19 +00:00
  • 117afeec91 (Auto)update libjingle 69188577-> 69260070 buildbot@webrtc.org 2014-06-16 07:11:01 +00:00
  • ab23d493e0 Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc. glaznev@webrtc.org 2014-06-13 23:31:35 +00:00
  • c6c1dfd7ea Add extra logging and latency restriction to VP8 HW encoder. glaznev@webrtc.org 2014-06-13 22:59:08 +00:00
  • a6764ab869 (Auto)update libjingle 69144530-> 69164179 buildbot@webrtc.org 2014-06-13 18:24:39 +00:00
  • af6f02f7bd Neon version of OverdriveAndSuppress() bjornv@webrtc.org 2014-06-13 14:50:23 +00:00
  • db56390f7e (Auto)update libjingle 69143161-> 69144530 buildbot@webrtc.org 2014-06-13 13:05:48 +00:00
  • f99c2f2dbc Add NACK feedback parameter to WebRtcVideoEngine2. pbos@webrtc.org 2014-06-13 12:27:38 +00:00
  • e322a175f6 Implement RTX tests+fixes in WebRtcVideoEngine2. pbos@webrtc.org 2014-06-13 11:47:28 +00:00
  • 9fbb717aca Remove engine_codecs_ cache from unittests. pbos@webrtc.org 2014-06-13 09:34:13 +00:00
  • d54ec1256c Fix GYP DEPTH for libjingle isolate files kjellander@webrtc.org 2014-06-13 09:16:23 +00:00
  • a1bfc50a72 Pass GYP DEPTH variable to isolate. kjellander@webrtc.org 2014-06-13 09:02:15 +00:00
  • c800c1cc40 (Auto)update libjingle 69131548-> 69132244 buildbot@webrtc.org 2014-06-13 07:56:17 +00:00
  • 1c8223c590 Initial owners file for talk/media/webrtc/. pbos@webrtc.org 2014-06-13 07:29:26 +00:00
  • 7e71b77f8a (Auto)update libjingle 69102234-> 69116997 buildbot@webrtc.org 2014-06-13 01:14:01 +00:00
  • 8e256eec4f Revert 6415 "Update generated asm offsets scripts." wu@webrtc.org 2014-06-12 23:03:17 +00:00
  • 1a6c6281ca Revert r6420 'Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck' Failing tests are disabled for memcheck. jiayl@webrtc.org 2014-06-12 21:59:29 +00:00
  • 3c13ed3b93 json.h include different header files depending on WEBRTC_CHROMIUM_BUILD being defined or not. Since json.h/cc is not even used in chromium it is the wrong flag to use. Instead add WEBRTC_EXTERNAL define. Also added OWNERS for base which is a copy of system_wrappers owners as the two folders are being merged. henrike@webrtc.org 2014-06-12 21:43:04 +00:00
  • ddeec048c0 Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck jiayl@webrtc.org 2014-06-12 21:42:46 +00:00
  • 3f3f428d2b (Auto)update libjingle 69097619-> 69099564 buildbot@webrtc.org 2014-06-12 21:36:26 +00:00
  • 6c6f33b5bb Fix the flaky RTP DataChannel test. jiayl@webrtc.org 2014-06-12 21:05:19 +00:00
  • 18dfa8d574 (Auto)update libjingle 69069003-> 69082899 buildbot@webrtc.org 2014-06-12 18:11:02 +00:00
  • cb254aac3b Enable pacing by default and remove the option to disable it from the new API. stefan@webrtc.org 2014-06-12 15:12:25 +00:00
  • 27f062ae6f Update generated asm offsets scripts. fgalligan@google.com 2014-06-12 14:59:28 +00:00
  • 4cb012858f Fixed GetStats when local and remote track are using the same ssrc. xians@webrtc.org 2014-06-12 14:57:05 +00:00
  • 7e3d62d709 Revert 6411 "Revert 6407 "Revert 6405 "Update generated asm offs..." kjellander@webrtc.org 2014-06-12 11:07:07 +00:00
  • b90619c07f (Auto)update libjingle 69049090-> 69054765 buildbot@webrtc.org 2014-06-12 09:19:08 +00:00
  • c01cc3d3a8 Revert 6407 "Revert 6405 "Update generated asm offsets scripts."" minyue@webrtc.org 2014-06-12 08:48:34 +00:00
  • 2881ab1e36 Increased kMaxRampUpDelayMs (120 to 240s). Add support for triggering on encode rsd metric if its thresholds are configured. Added unit tests. asapersson@webrtc.org 2014-06-12 08:46:46 +00:00
  • 276637b107 Disable flaky test on DrMemory Full. pbos@webrtc.org 2014-06-12 08:46:21 +00:00
  • d41eaeb7cd (Auto)update libjingle 69005149-> 69049090 buildbot@webrtc.org 2014-06-12 07:13:26 +00:00
  • 286cd7683c Revert 6405 "Update generated asm offsets scripts." henrike@webrtc.org 2014-06-12 00:38:32 +00:00
  • e9e8007ab4 (Auto)update libjingle 68985065-> 69005149 buildbot@webrtc.org 2014-06-11 18:41:17 +00:00
  • 4aeb94186a Update generated asm offsets scripts. fgalligan@google.com 2014-06-11 17:12:51 +00:00
  • 5b111b06fa Re-land "Create a joint encoder/decoder wrapper for iSAC in ACM" henrik.lundin@webrtc.org 2014-06-11 14:37:21 +00:00
  • 8454ad1b3e Reland: Making WebRTC able to play and record audio to files for tests. phoglund@webrtc.org 2014-06-11 14:12:04 +00:00
  • ab85187e63 Remove unused resource henrik.lundin@webrtc.org 2014-06-11 13:59:44 +00:00
  • 9e65a3b013 Re-land webrtcmediaengine.cc part of r6397. pbos@webrtc.org 2014-06-11 13:42:37 +00:00
  • fbb567dacd Add APIs to enable padding with redundant payloads. stefan@webrtc.org 2014-06-11 13:41:36 +00:00
  • 5d223a7d2d (Auto)update libjingle 68982444-> 68983526 buildbot@webrtc.org 2014-06-11 13:05:05 +00:00
  • 6604c6df26 Revert 6397 "(Auto)update libjingle 68949184-> 68982444" minyue@webrtc.org 2014-06-11 13:02:36 +00:00
  • af214d804f (Auto)update libjingle 68949184-> 68982444 buildbot@webrtc.org 2014-06-11 12:46:49 +00:00
  • e08a11c4a1 Revert 6395 "Making WebRTC able to play and record audio to file..." minyue@webrtc.org 2014-06-11 10:40:30 +00:00
  • fa042ca15d Making WebRTC able to play and record audio to files for tests. phoglund@webrtc.org 2014-06-11 09:57:23 +00:00
  • c726b1fc33 Revert r6377 "Create a joint encoder/decoder wrapper for iSAC in ACM" henrik.lundin@webrtc.org 2014-06-11 08:35:53 +00:00
  • 18026abd82 common_audio/signal_processing: Removes macro WEBRTC_SPL_RSHIFT_U16 bjornv@webrtc.org 2014-06-11 06:53:20 +00:00
  • 782978cfcb common_audio/signal_processing: Moves WEBRTC_SPL_UMUL_16_16_RSFT16 to iSAC fix bjornv@webrtc.org 2014-06-11 06:39:03 +00:00
  • 3f83072c26 modules/audio_processing: Adds a config for reported delays bjornv@webrtc.org 2014-06-11 04:48:11 +00:00
  • e61b8e32d8 Adds end to end DataChannel tests. jiayl@webrtc.org 2014-06-10 23:54:13 +00:00
  • a40210aee2 Add support for NVidia VP8 HW encoder. glaznev@webrtc.org 2014-06-10 23:48:29 +00:00
  • fd59c39caa Delete last file in neteq4 folder henrik.lundin@webrtc.org 2014-06-10 20:26:27 +00:00
  • 919914d71b MIPS optimizations for ISAC (patch #1) andrew@webrtc.org 2014-06-10 18:13:15 +00:00
  • 0d7ab0a634 Adding the new video folder and pacer to the wathclist. mflodman@webrtc.org 2014-06-10 13:59:37 +00:00
  • 12cd443752 Noise suppression: Change signature to work on floats instead of ints kwiberg@webrtc.org 2014-06-10 11:13:09 +00:00
  • 1014101470 Revert 6380 "Replace libjingle_root with talk_root variable." kjellander@webrtc.org 2014-06-10 10:13:38 +00:00
  • 3eb2c2f4c3 (Auto)update libjingle 68891947-> 68893961 buildbot@webrtc.org 2014-06-10 09:39:06 +00:00
  • 86f613d6b8 Move WebRtcVideoEngine2 fakes to unittest header. pbos@webrtc.org 2014-06-10 08:53:05 +00:00
  • 734a532723 Add additional metric (relative standard deviation of encode time) for overuse detection. This code is currently only for testing. asapersson@webrtc.org 2014-06-10 06:35:22 +00:00
  • 0238682984 Replace libjingle_root with talk_root variable. kjellander@webrtc.org 2014-06-10 05:46:31 +00:00
  • 7b82c18979 Add kjellander@webrtc.org as OWNER for *.isolate kjellander@webrtc.org 2014-06-10 05:42:53 +00:00
  • 620048172c Create a joint encoder/decoder wrapper for iSAC in ACM henrik.lundin@webrtc.org 2014-06-09 18:39:00 +00:00
  • a90abdef62 Add thread annotations to AcmReceiver henrik.lundin@webrtc.org 2014-06-09 18:35:11 +00:00
  • 190a32fd55 Make some methods in Clock class const declared henrik.lundin@webrtc.org 2014-06-09 17:40:49 +00:00
  • 6b6e58d632 Remove unused test_env.py from isolate files + fix nss path. kjellander@webrtc.org 2014-06-09 14:35:09 +00:00
  • 85d2794e5b Adds support for the "apt" format parameter and turns on the RTX feature. stefan@webrtc.org 2014-06-09 12:51:39 +00:00
  • ed7edb8e89 Enables DelayCorrection tests bjornv@webrtc.org 2014-06-09 10:02:05 +00:00
  • 582367f251 Updated conformance tests and w3c-ified them. phoglund@webrtc.org 2014-06-09 09:47:44 +00:00
  • a1a2c0c190 Multi-threaded unit test for Audio Coding Module using iSAC henrik.lundin@webrtc.org 2014-06-09 09:37:17 +00:00