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  • b2b628d5cd Further relax thresholds in mixing test. andrew@webrtc.org 2013-01-02 18:50:13 +00:00
  • 00c7c4315b Replace voice engine utility functions with system wrapper variants. andrew@webrtc.org 2013-01-02 16:06:39 +00:00
  • 943770b1ab Fixed various problems with the reformat script: * Improved sort checker so we sort includes more. * Fixed vars in brackets and varsLikeTHIS. * Added automatic x++ to ++x conversion in for loops. phoglund@webrtc.org 2013-01-02 15:46:43 +00:00
  • ec9c942e45 Reformatted thread and static_instance. phoglund@webrtc.org 2013-01-02 08:45:03 +00:00
  • a19d04e707 Coverity now uses Visual Studio 2010 project file kjellander@webrtc.org 2012-12-22 16:48:48 +00:00
  • 1b6da28047 Bugfix for NACK behavior. Current code sends a number of duplicate NACK requests. pwestin@webrtc.org 2012-12-21 17:46:24 +00:00
  • f556890844 Added possibility to repeat frames. Also added unittest for that feature. brykt@google.com 2012-12-20 11:42:45 +00:00
  • d73527ccab Changed assert to log. mflodman@webrtc.org 2012-12-20 09:26:17 +00:00
  • d0d41498a3 Adding AUDIO application as default for Opus stereo tina.legrand@webrtc.org 2012-12-20 09:23:10 +00:00
  • ad0ed582b5 Fixed a missed initialization (found by valgrind FYI bot). phoglund@webrtc.org 2012-12-20 09:14:36 +00:00
  • ac77084583 Roll opus to 172355 and delete opus_demo from webrtc opus leozwang@webrtc.org 2012-12-19 17:24:30 +00:00
  • 6bc5d4dc07 Reformatted sort. phoglund@webrtc.org 2012-12-19 14:55:24 +00:00
  • 1960219530 Make protection method, filename and resolution configurable for FullStackTest. stefan@webrtc.org 2012-12-19 12:45:16 +00:00
  • 4275ab1ca0 Implement NetEq duration estimation for Opus. tina.legrand@webrtc.org 2012-12-19 09:52:45 +00:00
  • 515ef2428c Clean up variable after it gets deleted leozwang@webrtc.org 2012-12-19 05:36:36 +00:00
  • e239bf0940 Making I420VideoFrame ref-counted mikhal@webrtc.org 2012-12-19 00:07:57 +00:00
  • b13dfbffd7 Making barcode tools work on Windows + fixes. kjellander@webrtc.org 2012-12-18 19:53:00 +00:00
  • 0b18fb38e6 vie auto test: Adding a constructor for NetworkParameters mikhal@webrtc.org 2012-12-18 19:47:52 +00:00
  • 622c8bd0cc ViE autotest: Adding loss models to the external transport mikhal@webrtc.org 2012-12-18 17:21:51 +00:00
  • 6e0ce73741 Reformatted map classes. phoglund@webrtc.org 2012-12-18 17:18:35 +00:00
  • 61f39a3174 Fixed bad header name. phoglund@webrtc.org 2012-12-18 16:02:13 +00:00
  • 07bf43c673 Replaced the _audio parameter with a strategy. phoglund@webrtc.org 2012-12-18 15:40:53 +00:00
  • 59ad541e57 Reformatted rw_lock classes. phoglund@webrtc.org 2012-12-18 15:20:35 +00:00
  • eaebeb36ae Without specifying the input files the offsets will not automatically be regenerated when building for different architectures. That is very risky as it will cause crashes rather than build errors. stefan@webrtc.org 2012-12-18 08:38:50 +00:00
  • 10abe25f6d Make audioproc output files be written to output dir by default. kjellander@webrtc.org 2012-12-17 18:28:07 +00:00
  • 3c37354b70 Initialize 3 variables which are preventing VS2012 from building. BUG=1211 TESTED=ninja -C out\Release Review URL: https://webrtc-codereview.appspot.com/992005 fbarchard@google.com 2012-12-15 01:09:18 +00:00
  • 4c32439830 Roll libyuv to r520. Includes security fix to mark stack as not executable. BUG=1172 TEST=none Review URL: https://webrtc-codereview.appspot.com/1000005 fbarchard@google.com 2012-12-15 00:20:08 +00:00
  • ad6845f4c4 Updated version number to 3.19 TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/995007 elham@webrtc.org 2012-12-14 21:28:09 +00:00
  • c5fcb0879b Update trace_event.h to match the one in Chromium hclam@chromium.org 2012-12-14 21:16:46 +00:00
  • dec09eed2f libyuv r515 ports matrix effects to Neon BUG=none TESTED=try bots Review URL: https://webrtc-codereview.appspot.com/966034 fbarchard@google.com 2012-12-14 15:22:25 +00:00
  • 4aee6b637d Added API to get receive side video delay. mflodman@webrtc.org 2012-12-14 14:02:10 +00:00
  • 1c75918302 Disabled flaky test. phoglund@webrtc.org 2012-12-14 10:40:05 +00:00
  • 7659d914bb Decoupled video rtp receiver from rtp receiver. phoglund@webrtc.org 2012-12-14 09:57:37 +00:00
  • 52d981f60c Reformatted list classes. phoglund@webrtc.org 2012-12-14 09:52:34 +00:00
  • 32519398b6 Remove latency excl network and add render time diff stats. stefan@webrtc.org 2012-12-14 09:03:27 +00:00
  • b8ba4d8109 Add number of inserted samples to NetEq statistics. roosa@google.com 2012-12-14 00:06:18 +00:00
  • c454fab03b Reformatting ACM. All changes are bit-exact in this CL. turaj@webrtc.org 2012-12-13 22:46:43 +00:00
  • ddebc17bee Fix for buffer overflow, WebRTC issue 1196 Review URL: https://webrtc-codereview.appspot.com/998004 elham@webrtc.org 2012-12-13 21:55:47 +00:00
  • 96dc6270d4 vpm unit test: Diasble frame dropping in tests mikhal@webrtc.org 2012-12-13 19:53:26 +00:00
  • 4493db5a3e vpm: removing unnecessary memcpy mikhal@webrtc.org 2012-12-13 18:25:36 +00:00
  • 7acb65a870 Added jitter to fake network pipe. mflodman@webrtc.org 2012-12-13 15:53:11 +00:00
  • 91c91df35a Track the actual render time rather than the decode time. stefan@webrtc.org 2012-12-13 15:26:01 +00:00
  • e19b078ebe Changed so that frame_cutter takes and argument where one can specify in which interval the frames should be deleted between the first frame to cut and the last frame to cut. This can for example be used to decrease the frame rate. brykt@google.com 2012-12-13 14:46:40 +00:00
  • 0240e8e90f Wider TSAN suppression for issue 300 kjellander@webrtc.org 2012-12-13 13:02:29 +00:00
  • 92bb417cb1 Decoupled RTP audio processor from RTP receiver. phoglund@webrtc.org 2012-12-13 10:48:24 +00:00
  • 5b689efe8e Will now only require near-perfect PSNR and SSIM. phoglund@webrtc.org 2012-12-13 10:15:06 +00:00
  • 86464eacb6 ISAC_main_inst initialized to NULL to avoid potentially garbage pointer passed to WebRtcIsacfix_EncoderInit BUG=1211 TESTED=local build on Windows. Failed previously with vs2012. With this change kenny.cc builds. Review URL: https://webrtc-codereview.appspot.com/984004 fbarchard@google.com 2012-12-13 07:47:54 +00:00
  • a8544eaf03 Vp8 tests: Removing legacy unused tests and reorganization of existing ones. mikhal@webrtc.org 2012-12-13 00:37:22 +00:00
  • 7877b0f6d2 Added noexecstack markers for assembly files (webrtc issue 1172). Webrtc builds on ios, linux, android and other major platforms passed. Didn't do chrome build test. Review URL: https://webrtc-codereview.appspot.com/987004 kma@webrtc.org 2012-12-12 23:22:13 +00:00
  • fa5b6bf4f4 Optimized WebRtcIsacfix_Spec2Time() for iSAC-Fix in ARM Neon processor. Speed doubled. Review URL: https://webrtc-codereview.appspot.com/930033 kma@webrtc.org 2012-12-12 23:00:52 +00:00
  • 1b60ceb499 Add GetAudioFrame API to VoiceEngine. roosa@google.com 2012-12-12 23:00:29 +00:00
  • b718619f0a Expose NetEq playout mode off through VoiceEngine. roosa@google.com 2012-12-12 21:59:14 +00:00
  • 0870f02cdb Add API to retreive last received RTP timestamp to VoiceEngine. roosa@google.com 2012-12-12 21:31:41 +00:00
  • d8aeb30d55 Revert 3269 > Will now only require near-perfect PSNR and SSIM. > > BUG= > TEST=Ran test and checked we accept somewhat lower values. > > Review URL: https://webrtc-codereview.appspot.com/964031 andrew@webrtc.org 2012-12-12 20:58:32 +00:00
  • 735a6cec96 Will now only require near-perfect PSNR and SSIM. phoglund@webrtc.org 2012-12-12 15:20:34 +00:00
  • 740be44af5 Reformatted file_* classes. phoglund@webrtc.org 2012-12-12 12:52:15 +00:00
  • 4e16f25774 Remove atomicops.h from WebRTC hclam@chromium.org 2012-12-12 01:13:19 +00:00
  • 9f0fc97d2f Rolllibvpx to 7a09f6b89268 marpan@webrtc.org 2012-12-11 23:02:32 +00:00
  • 770a01e3b0 Fix build by including trace_event_internal in webrtc namespace hclam@chromium.org 2012-12-11 22:51:38 +00:00
  • f222a00881 Use TRACE_EVENT to track time spent in VP8 encoding hclam@chromium.org 2012-12-11 22:27:55 +00:00
  • d2bcde2e4e Suppressing TSan warnings for system_wrappers_unittests kjellander@webrtc.org 2012-12-11 21:39:18 +00:00
  • ad7efa6944 Port Chromium's trace_event.h to WebKit and add hclam@chromium.org 2012-12-11 21:19:08 +00:00
  • 02d9df4544 Updated webrtc_resources_revision to 11, for adding two test files for APM and iSAC. Review URL: https://webrtc-codereview.appspot.com/973014 kma@webrtc.org 2012-12-11 19:33:56 +00:00
  • 71258c594b Add a third full stack test and support for random jitter in ext transport. stefan@webrtc.org 2012-12-11 15:14:56 +00:00
  • eaf7cf26fe Adding a simple fake network pipe to use for testing. Next CL will contain an external transport implementation using this link and I'll follow up later making this more advanced. mflodman@webrtc.org 2012-12-11 11:47:22 +00:00
  • f98ffc6db3 Removing default trybot names kjellander@webrtc.org 2012-12-11 10:00:41 +00:00
  • 42259e7ebc VoE Changes to enable dual_streaming. turaj@webrtc.org 2012-12-11 02:15:12 +00:00
  • 36965b1803 Bug fix for iSAC fixed-point. The bug was the result of changes in iSAC floating-point to add 48 kHz extension. turaj@webrtc.org 2012-12-10 23:52:43 +00:00
  • 55edaecc93 Revert r3254 due to bot failure on android. marpan@webrtc.org 2012-12-10 19:59:03 +00:00
  • 1f3476dd83 Roll libvpx to 000c8414b510. marpan@webrtc.org 2012-12-10 18:11:30 +00:00
  • 5bbe069f28 Reformatted event* classes. phoglund@webrtc.org 2012-12-10 10:44:37 +00:00
  • 3bb42ef0d6 Made e2e audio quality test write its results to perf. phoglund@webrtc.org 2012-12-10 10:08:00 +00:00
  • 72feb0b2e2 Not to enum NOTPRESENT audio devices with CoreAudio on Win braveyao@webrtc.org 2012-12-07 22:36:07 +00:00
  • 8e49b02f3d Add more audio codec information into codec list leozwang@webrtc.org 2012-12-07 22:26:57 +00:00
  • 451aa5dd9d Adding vp8 sequence coder: simple command line encode and decode. Goal is to replace existing normal test and affiliates (will be done in follow up cl's) BUG =1070 mikhal@webrtc.org 2012-12-07 21:23:02 +00:00
  • 3a5a8a8bcc Properly zero out unmixed frames. andrew@webrtc.org 2012-12-07 19:37:16 +00:00
  • 0e739508e0 Added buildbot benchmarking in iSAC and APM into Android platform build. Review URL: https://webrtc-codereview.appspot.com/964022 kma@webrtc.org 2012-12-07 15:26:28 +00:00
  • b968213f3c vp8 test: Updating creation of enc/dec mikhal@webrtc.org 2012-12-06 19:14:26 +00:00
  • 251f64e9e8 Updating vp8 test structure mikhal@webrtc.org 2012-12-06 17:56:20 +00:00
  • 60d25f90ff Updating Vp8 unit tests - Initiating the switch to gtest-based tests, and adding a stride test. This is a follow up on r3227. mikhal@webrtc.org 2012-12-06 16:48:18 +00:00
  • 75f8c78d08 Fixing path to ptypes.txt in NetEqRTPplay henrik.lundin@webrtc.org 2012-12-06 15:22:00 +00:00
  • df94329a4b Use different cpufeatures library when building with chrome. wjia@webrtc.org 2012-12-05 22:54:52 +00:00
  • 81cffd1f2c Port Chromium's atomicops to WebRTC hclam@chromium.org 2012-12-05 22:51:01 +00:00
  • 63a243a031 Replace the last occurrence of .s with .h leozwang@webrtc.org 2012-12-05 07:12:15 +00:00
  • 96bcac8fbb Expose Set and Get Recording/Playout sample rate apis leozwang@webrtc.org 2012-12-04 19:11:55 +00:00
  • f4e070eca5 Added auto-call feature to WebRTCDemo. fischman@webrtc.org 2012-12-04 16:53:43 +00:00
  • 2cf22a6abc Revert 3231 - VoE Changes to enable dual_streaming. perkj@webrtc.org 2012-12-04 10:02:02 +00:00
  • e861359925 Adds two full stack performance metrics for end-to-end delay. stefan@webrtc.org 2012-12-04 09:40:46 +00:00
  • 6bd737a714 First pass of MediaCodecDecoder which uses Android MediaCodec API. dwkang@webrtc.org 2012-12-04 06:38:19 +00:00
  • 781cf06124 libyuv r508 with scaler fix for overread horizontally that was caught by valgrind. BUG=none TEST=valgrind bots Review URL: https://webrtc-codereview.appspot.com/968013 fbarchard@google.com 2012-12-04 03:07:29 +00:00
  • 767d87cf24 VoE Changes to enable dual_streaming. turaj@webrtc.org 2012-12-03 22:51:37 +00:00
  • 226db898f7 Dual-stream implementation, not including VoE APIs. Review URL: https://webrtc-codereview.appspot.com/933015 turaj@webrtc.org 2012-12-03 22:13:31 +00:00
  • 277ec8e3f5 Fix a bug when iSAC-48kHz was added. turaj@webrtc.org 2012-12-03 21:16:23 +00:00
  • f18de86db1 Revert 3227 > vp8 unittest: Adding qcif stride test > > Review URL: https://webrtc-codereview.appspot.com/930030 mikhal@webrtc.org 2012-12-03 20:08:57 +00:00
  • ab83bb39ad vp8 unittest: Adding qcif stride test mikhal@webrtc.org 2012-12-03 19:12:29 +00:00
  • b0dff12d2b 48 kHz extension to iSAC. turaj@webrtc.org 2012-12-03 17:43:52 +00:00
  • 0bacb635cb Removed stale version of fuzzer; it's now internal. phoglund@webrtc.org 2012-12-03 14:28:40 +00:00
  • 8d0cd07d0c Add test to verify that padding only frames are passing through the RTP module. stefan@webrtc.org 2012-12-03 14:01:46 +00:00
  • 5b4fe494e7 Changing default bitrate to 64000 bps for Opus. tina.legrand@webrtc.org 2012-12-03 12:08:53 +00:00
  • ad0f3baf90 Removing redundant codec unittest targets. kjellander@webrtc.org 2012-12-03 10:52:29 +00:00