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  • ac1f877a5e Remove unneeded libvpx path from vp8 include_dirs. andrew@webrtc.org 2013-02-25 17:10:38 +00:00
  • 9ae1354e25 Refactor ring_buffer interface, add a feature and a test. andrew@webrtc.org 2013-02-25 17:07:35 +00:00
  • 8a0662306d New attempt at fixing hard-coded libvpx source. phoglund@webrtc.org 2013-02-25 15:05:01 +00:00
  • 9a6623b629 Revert "Fixing hard-coded libvpx source path." phoglund@webrtc.org 2013-02-25 14:27:14 +00:00
  • 8571c90c7f Fixing hard-coded libvpx source path. phoglund@webrtc.org 2013-02-25 14:09:12 +00:00
  • 2f9bd247ad Ported assembly coding in APM from Android to iOS. Bugs=none Test=trybots, and offline file bit-exact tests. Review URL: https://webrtc-codereview.appspot.com/1066009 kma@webrtc.org 2013-02-23 04:16:59 +00:00
  • 0d8d010017 Handle multiple calls to set initial delay mikhal@webrtc.org 2013-02-22 19:30:44 +00:00
  • 7bf7326d0b Remove WEBRTC_TRACE completely when tracing is disabled. wjia@webrtc.org 2013-02-22 00:54:58 +00:00
  • 10987a851d Minor bug fix in maxFPS parameter declaration. vikasmarwaha@webrtc.org 2013-02-21 22:03:39 +00:00
  • bf3a9b3cce Fix for WebRTC Issue 1384. Some cameras return 0 fps for all capabilities which causes divide-by-zero. Review URL: https://webrtc-codereview.appspot.com/1101013 vikasmarwaha@webrtc.org 2013-02-21 20:25:54 +00:00
  • 5140e24037 MIPS optimizations for Signal Processing Library patch01 andrew@webrtc.org 2013-02-21 20:12:21 +00:00
  • 60f83131e4 AEC refactoring: Moved typedefs to _internal.h bjornv@webrtc.org 2013-02-21 16:12:24 +00:00
  • 7a7a008031 Changing non-const reference arguments to pointers, ACM tina.legrand@webrtc.org 2013-02-21 10:27:48 +00:00
  • f61e02c81f Misc cleanups to webrtc/android code: - Replace some deprecated calls/enums with their more modern equivalents. - Clean up some usage of global data and/or hide it better - Catch specific exceptions instead of Exception, and log the exception instead of just its message. - Random log message cleanups - Added a build_with_libjingle gyp variable to mimic build_with_chromium for when webrtc is built as part of a libjingle project but not part of chromium. fischman@webrtc.org 2013-02-20 23:13:46 +00:00
  • 56a9ec30e9 Refactoring AEC: AecCore struct made private bjornv@webrtc.org 2013-02-20 22:38:47 +00:00
  • 71e91f3b64 Refactor AEC: PowerLevel bjornv@webrtc.org 2013-02-20 19:24:50 +00:00
  • 4d1cfae622 Added a pointer getter to the system_delay variable. Tested with audioproc_unittest, trybots bjornv@webrtc.org 2013-02-20 17:31:38 +00:00
  • 47b274de44 Refactoring AEC: Added a SetConfigCore function bjornv@webrtc.org 2013-02-20 17:09:47 +00:00
  • 716fd90ff2 Moved out buffer handling to ProcessFrame() bjornv@webrtc.org 2013-02-20 16:59:41 +00:00
  • ee7202f7a4 Removed unused get_config function. The configuration is already stored and handled in the audio processing module, so there is no need for this functionality. bjornv@webrtc.org 2013-02-20 16:36:59 +00:00
  • 59b2d5fbce Stop and restart fix. mflodman@webrtc.org 2013-02-20 16:00:27 +00:00
  • eb7ebf20ed Revert 3543 > Changing non-const reference arguments to pointers, ACM > > Part of refactoring of ACM, and recent lint-warnings. > This CL changes non-const references in the ACM API to pointers. > > BUG=issue1372 > > Review URL: https://webrtc-codereview.appspot.com/1103012 tina.legrand@webrtc.org 2013-02-20 15:57:31 +00:00
  • 374aa49e1a Changing non-const reference arguments to pointers, ACM tina.legrand@webrtc.org 2013-02-20 15:22:23 +00:00
  • 0b6293aaaa Fixed typo in vie_autotest_loopback.cc. pbos@webrtc.org 2013-02-20 12:13:10 +00:00
  • 83663efba4 Replace gtest_prod.h include with our own FRIEND_TEST macro. andrew@webrtc.org 2013-02-19 23:43:57 +00:00
  • aea96d36e3 Rename webrtc::StatsObserver to webrtc::CallStatsObserver to avoid ODR violations with peerconnectioninterface.h in libjingle. fischman@webrtc.org 2013-02-19 22:09:36 +00:00
  • 0a480cbe4d Added getter for far_time_buf in AEC. Only used in AEC debug dump. bjornv@webrtc.org 2013-02-19 21:41:27 +00:00
  • 5fc829200c This refactoring CL moves the nlp_mode member value from aecpc_t to aec_t, since it it never used at that level. Further, I removed two suppression variables by depending on nlp_mode directly. bjornv@webrtc.org 2013-02-19 21:06:52 +00:00
  • cea70f4055 * Name change * Removed WebRtcAec_ function name prepending on private function. bjornv@webrtc.org 2013-02-19 21:03:10 +00:00
  • 95b48c3551 Update to codec unit test: enable frame dropper for rate control test. Review URL: https://webrtc-codereview.appspot.com/1099014 marpan@webrtc.org 2013-02-19 20:02:32 +00:00
  • 77fced32e2 fixing nack list size calculation mikhal@webrtc.org 2013-02-19 19:42:35 +00:00
  • 10741b32b8 Updated version number to 3.24 Review URL: https://webrtc-codereview.appspot.com/1110004 elham@webrtc.org 2013-02-19 19:05:14 +00:00
  • 1682f71210 Updating watchlist mikhal@webrtc.org 2013-02-19 18:31:20 +00:00
  • ba23d11564 Will now run pylint on all python files if there's at least one modified python file in the checkin. phoglund@webrtc.org 2013-02-19 12:27:57 +00:00
  • 0460c7294a Remove the dependency on dxguid.lib. It turns out we don't really need it and therefore can also get rid of the added lib directory. Review URL: https://webrtc-codereview.appspot.com/1094015 tommi@webrtc.org 2013-02-19 12:13:03 +00:00
  • d2c3bed1da Move directx_sdk_path definition variable into the video_render_module gyp file. tommi@webrtc.org 2013-02-18 14:53:04 +00:00
  • eb91792cfd Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings. stefan@webrtc.org 2013-02-18 14:40:18 +00:00
  • 3897255b63 Add VoE interface to VieRTP test mikhal@webrtc.org 2013-02-16 01:35:59 +00:00
  • e3d6ffede4 Increase threshold in codec unit test. Review URL: https://webrtc-codereview.appspot.com/1096011 marpan@webrtc.org 2013-02-15 23:31:17 +00:00
  • ef9f76a59d Adding a receive side API for buffering mode. At the same time, renaming the send side API. mikhal@webrtc.org 2013-02-15 23:22:18 +00:00
  • 47fe5736c1 Bug fix for webrtc issue 1391. Typo in sin_length for socket address. Review URL: https://webrtc-codereview.appspot.com/1108004 vikasmarwaha@webrtc.org 2013-02-15 18:42:12 +00:00
  • b4cd342eb9 This refactoring CL contains an API to get low level echo metrics stats. bjornv@webrtc.org 2013-02-15 18:40:34 +00:00
  • 21a2fc902d This Cl includes * A getter for echo_state * Style changes, such as changes to int where appropriate bjornv@webrtc.org 2013-02-15 17:01:03 +00:00
  • 325f625137 Moved the actual calculations to aec_core to avoid passing up low level members. bjornv@webrtc.org 2013-02-15 15:21:02 +00:00
  • 0989fb7bfa Make VoiceEngineImpl inherit from VoiceEngine. This associates the two types instead of incorrectly reinterpret casting VoiceEngineImpl* to VoiceEngine* (since these types were previously unrelated). tommi@webrtc.org 2013-02-15 15:07:32 +00:00
  • 17238576ba Removed astyle from webrtc_reformat since clang-format-chrome.py handles that now. phoglund@webrtc.org 2013-02-15 09:43:20 +00:00
  • 076fc12539 Modify SincResampler to build in webrtc. andrew@webrtc.org 2013-02-15 03:54:22 +00:00
  • 6f6acd9f80 Duplicated sampling frequency multiplier to aecpc_t struct to avoid a getter. bjornv@webrtc.org 2013-02-14 21:17:12 +00:00
  • 4013ac478e Roll Chromium revision 176094:182149 kjellander@webrtc.org 2013-02-14 19:13:30 +00:00
  • 7267ffde56 Moved debug file handling to aec_core from echo_cancellation.c. This removes dependency on low level member variables. bjornv@webrtc.org 2013-02-14 17:56:23 +00:00
  • 3e10249f20 Added delay estimation test to audio processing unit tests. bjornv@webrtc.org 2013-02-14 15:29:09 +00:00
  • e580be993c Add regression monitoring for audioproc and iSAC fixed-point tests. kjellander@webrtc.org 2013-02-14 12:27:17 +00:00
  • 07b667db5e Remove MultiStreamMode from test. stefan@webrtc.org 2013-02-14 11:35:20 +00:00
  • 294e5b0b82 Reset ssrc when calling SetSendCodec. mflodman@webrtc.org 2013-02-14 11:25:26 +00:00
  • a092cbf9b7 Fixing lint warnings from previous commit tina.legrand@webrtc.org 2013-02-14 09:28:10 +00:00
  • 45eab19e7d Import stringize_macros from Chromium. andrew@webrtc.org 2013-02-13 23:37:14 +00:00
  • a8ef811fe5 Import SincResampler from Chromium. andrew@webrtc.org 2013-02-13 23:00:49 +00:00
  • 9c4e662ea8 Fix Windows x64 errors in video_codecs_test_framework kjellander@webrtc.org 2013-02-13 09:35:12 +00:00
  • 6388c3e2fd Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM. turaj@webrtc.org 2013-02-12 21:42:18 +00:00
  • e6e344a7dc Sync libvpx and its gyp wrapper from Chromium. andrew@webrtc.org 2013-02-12 19:35:18 +00:00
  • 0ee57c2436 Increase maximum resolution to 4k x 3k. BUG=1375 TESTED=try bots Review URL: https://webrtc-codereview.appspot.com/1097008 fbarchard@google.com 2013-02-12 04:57:56 +00:00
  • 57a0049e25 VCM: Removing frame drop enable from Reset call BUG = 1387 mikhal@webrtc.org 2013-02-11 21:23:23 +00:00
  • 18a21a03c6 Android NDK build tools kjellander@webrtc.org 2013-02-11 17:43:19 +00:00
  • 00ab7cf4fd Fix perf output for audioproc and iSAC fixed-point tests kjellander@webrtc.org 2013-02-11 12:33:03 +00:00
  • 0cb48a0a18 Set SingleStream BWE in unittests. stefan@webrtc.org 2013-02-11 08:30:23 +00:00
  • 63066f7200 Set qpMax to 56 in for all VP8 tests. Fixes buildbot breakage. stefan@webrtc.org 2013-02-11 08:27:33 +00:00
  • 3d305c64b4 Updates to send side streaming mode: 1. Disabling frame-droppers from the vie encoder and not the channel. 2. Accounting for qpMax in the VP8 wrapper. mikhal@webrtc.org 2013-02-10 18:42:55 +00:00
  • 79481474ad Update version number to 3.23 tnakamura@webrtc.org 2013-02-08 19:56:09 +00:00
  • 7c850745d3 Adding third_party/directx and winsdk_samples to svn:ignore kjellander@webrtc.org 2013-02-07 16:12:25 +00:00
  • 687efe3ba8 Adding third_party/opus to svn:ignore kjellander@webrtc.org 2013-02-07 11:16:09 +00:00
  • b64732abfc Fix Win64 build breakage henrikg@webrtc.org 2013-02-07 10:14:05 +00:00
  • 147c73ea60 Made it possible to render custom call output to file. phoglund@webrtc.org 2013-02-07 08:52:08 +00:00
  • d83b9fdf45 Fixed a bug in iSAC transform functions on ARM-Neon platform. Performance unchanged. Bugs=none Test=trybots, and file bit-exact tests; passed. kma@webrtc.org 2013-02-06 23:53:13 +00:00
  • 4fd5527ab1 Don't report an error for GetEstimatedReceiveBandwidth if there is no valid estimate. mflodman@webrtc.org 2013-02-06 17:46:39 +00:00
  • fe3d606f15 Enable indefinitely running vie_auto_test option kjellander@webrtc.org 2013-02-06 09:36:37 +00:00
  • 1e7ed7afe9 Use LOG_F interface for unsupported functions. andrew@webrtc.org 2013-02-05 21:23:39 +00:00
  • 959da8d286 Added labels in transform_neon.S in iSAC-fix, so the tables be shared with other files in iOS build. Also, moved several code lines in the same file, in case register values cannot be preserved after a function call which could cause a crash in some platforms (e.g. iOS etc.). kma@webrtc.org 2013-02-05 20:46:55 +00:00
  • a7303bdfb5 Lint-cleaned video and audio receivers. phoglund@webrtc.org 2013-02-05 15:12:39 +00:00
  • c4e45f67c0 Updated version number to 3.22 Review URL: https://webrtc-codereview.appspot.com/1096004 elham@webrtc.org 2013-02-04 21:34:05 +00:00
  • 23e3559507 Updating Perf numbers for Win Large Test. tina.legrand@webrtc.org 2013-02-04 13:25:11 +00:00
  • 244251a9cd Moved almost all payload-related stuff to the payload registry. phoglund@webrtc.org 2013-02-04 13:23:07 +00:00
  • fa53d8717c Fixing/disabling Windows x64 warnings kjellander@webrtc.org 2013-02-04 10:07:17 +00:00
  • 254d85af54 Exchange TRY by enumerating image formats in Linux video capture braveyao@webrtc.org 2013-02-04 07:53:53 +00:00
  • 6ed8ebcef9 Fix MaxChannels test; 32 -> 100. andrew@webrtc.org 2013-02-02 00:05:58 +00:00
  • 4a6f62d4dc Remove (in practice) the voice engine channel limit. andrew@webrtc.org 2013-02-01 23:42:44 +00:00
  • dbe97d2550 Adding a send side API for streaming mikhal@webrtc.org 2013-02-01 19:33:21 +00:00
  • becf9c897c Fix mismatch between different NACK list lengths and packet buffers. stefan@webrtc.org 2013-02-01 15:09:57 +00:00
  • b586507986 Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages. stefan@webrtc.org 2013-02-01 14:33:42 +00:00
  • 46d90dcd74 Adding three frame sizes to Opus tina.legrand@webrtc.org 2013-02-01 14:20:06 +00:00
  • d087789b9c Adjusted net_50_5_plr_5 on Linux, removed all gilbert_elliot metrics (too flaky), added mac expectations. phoglund@webrtc.org 2013-02-01 13:28:58 +00:00
  • aaad6134b9 Implementing stereo support for G.722 henrik.lundin@webrtc.org 2013-02-01 11:49:28 +00:00
  • 7050f96bff Set frame length for frame converting in external renderer braveyao@webrtc.org 2013-02-01 02:49:29 +00:00
  • ac46c6dac3 Replaced relative path to reference from <(webrtc_root). bjornv@webrtc.org 2013-01-31 21:06:16 +00:00
  • 9d532fd275 Fix propagating RED paylaod-type to ACM. turaj@webrtc.org 2013-01-31 18:34:19 +00:00
  • 763faeab4e Removing a codec from NetEq database has a bug. |funcDurationEst| is not updated. turaj@webrtc.org 2013-01-31 18:21:58 +00:00
  • c0ada864b2 fix for issue 281. turaj@webrtc.org 2013-01-31 18:21:06 +00:00
  • 8c8ad85c5d fix issue 1322, accept -1 as default payload-type for redundant coding (FEC). turaj@webrtc.org 2013-01-31 18:20:17 +00:00
  • 119c67df36 Adding a max jitter filter to the JB estimate - allowing two modes, one will return the last estimate (current setting), and another will return the max value seen, and allow setting an initial value. This cl also includes tests and some clean up. mikhal@webrtc.org 2013-01-31 17:18:02 +00:00
  • e07c661a29 VP8: Making key frame interval a tunnable parameter mikhal@webrtc.org 2013-01-31 16:37:13 +00:00
  • 6e3968f62a Fix NetEq4 unit tests for VS2012 henrik.lundin@webrtc.org 2013-01-31 15:07:30 +00:00