Commit Graph

  • 99b1a32146 Retyped the frequency estimate of the comfort noise for the higher band to harmonize the AEC code. -Changed the type for the frequency estimate of the comfort noise for the higher band to be a two dimensional float array instead of a complex_t array. This makes sense since all the other frequency estimate (apart from the coherence) use this format and doing this change allows bundling the IFFT operations into using the InverseFFT method. -Moved the memset of the frequency estimate of the comfort noise to where it is used and made it conditional so that it is only performed when used. -Harmonized the if-statements for when the frequency estimate of the comfort noise is computed in the different optimized ComfortNoise computation methods. peah 2015-12-16 06:07:23 -08:00
  • 426ae9dc21 Roll chromium_revision 6e5b8cb..2c8eb1f (365419:365513) kjellander 2015-12-16 04:25:39 -08:00
  • a6db4958c9 Move Rent-A-Codec out of CodecManager kwiberg 2015-12-16 04:19:08 -08:00
  • a29386c26d Make VoiceDetection not a ProcessingComponent (bit exact). solenberg 2015-12-16 03:31:12 -08:00
  • 672aba3f57 Fix error prone code in VideoCapturerAndroid perkj 2015-12-16 02:17:16 -08:00
  • 66085beef8 Bugfix that fixes the error where the audio processing module is called using the wrong sample rate for the render signal. peah 2015-12-16 02:02:20 -08:00
  • 54999d411b rtcp::Dlrr block moved into own file and got Parse function danilchap 2015-12-16 01:56:20 -08:00
  • 29e2f9385b Fix NoiseSuppression initialization behavior. This was changed when removing the ProcessingComponent inheritance in https://codereview.webrtc.org/1507683006/. solenberg 2015-12-16 01:18:15 -08:00
  • 45fd9fe92c New macro: RTC_DEPRECATED (for annotating deprecated functions) kwiberg 2015-12-16 01:09:16 -08:00
  • ed644d8c07 Roll chromium_revision bff4606..6e5b8cb (365226:365419) kjellander 2015-12-15 20:30:14 -08:00
  • eb45981165 Restoring behavior where PeerConnection tracks changes to MediaStreams. deadbeef 2015-12-15 19:24:43 -08:00
  • 44f0819978 Fixing bug where "mid" wasn't preserved across re-offers. deadbeef 2015-12-15 16:20:09 -08:00
  • c1316a1e51 Fix HPF initialization behavior. This was changed when removing the ProcessingComponent inheritance in https://codereview.webrtc.org/1490333004/. solenberg 2015-12-15 16:07:25 -08:00
  • 95d9851a6c Add speech encoder to the encoder stack specification struct kwiberg 2015-12-15 14:21:33 -08:00
  • 7eb914debb Fix incorrect comment kwiberg 2015-12-15 14:20:24 -08:00
  • 78315b9813 Reland of Base webrtc fuzzers on a template. (patchset #1 id:1 of https://codereview.webrtc.org/1528043002/ ) Peter Boström 2015-12-15 21:57:52 +01:00
  • f9945b2d1a Only try to pair protocol matching candidates for creating connections. If the local port and the remote candidate's protocols do not match, do not even try to pair them. This avoids printing out confusing logs like "Attempt to change a remote candidate..." in p2ptransportchannel when two remote candidates have the same port number but different protocols. Honghai Zhang 2015-12-15 12:20:13 -08:00
  • 949028fbf1 Make LevelEstimation not a ProcessingComponent. solenberg 2015-12-15 11:39:38 -08:00
  • 5e0218c66e Revert of Base webrtc fuzzers on a template. (patchset #1 id:1 of https://codereview.webrtc.org/1524993002/ ) tommi 2015-12-15 10:23:55 -08:00
  • 51254331cc Android: Refactor renderers to allow apps to inject custom shaders Magnus Jedvert 2015-12-15 16:22:29 +01:00
  • 91941ae493 rtcp::VoipMetric block moved into own file and got Parse function danilchap 2015-12-15 07:06:36 -08:00
  • 32d989b3f2 Disable transport sequence numbers for audio. Stefan Holmer 2015-12-15 15:55:10 +01:00
  • 10aea22c84 Roll chromium_revision 53970fd..bff4606 (365141:365226) kjellander 2015-12-15 05:37:54 -08:00
  • 377b5e622f enabled cpplint for the webrtc/modules/rtp_rtcp directory danilchap 2015-12-15 04:33:44 -08:00
  • 6eca7e3c37 Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :( tommi 2015-12-15 04:27:11 -08:00
  • 6db6cdc604 [rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs danilchap 2015-12-15 02:54:47 -08:00
  • 9638143033 Reland of Made EglBase an abstract class and cleaned up. (patchset #1 id:1 of https://codereview.webrtc.org/1522073002/ ) perkj 2015-12-15 02:48:07 -08:00
  • e005cf2c93 [rtp_rtcp] SSRCDatabase class cleaned (including all lint errors) danilchap 2015-12-15 01:59:47 -08:00
  • 5ea3da2cbb Base webrtc fuzzers on a template. Peter Boström 2015-12-15 10:46:13 +01:00
  • 8f09f170e6 Simple CL to fix lint errors in webrtc/modules/remote_bitrate_estimator. Added the lint check for the folder to the presubmit script. terelius 2015-12-15 00:51:54 -08:00
  • 498ae00f39 Disable ThreadTest.ThreeThreadsInvoke on DrMemory bots. Stefan Holmer 2015-12-15 09:46:06 +01:00
  • 47a740bc5e [rtp_rtcp] lint errors about rand() usage fixed. danilchap 2015-12-15 00:30:07 -08:00
  • 2d36b9233e Roll chromium_revision 10bf0e1..53970fd (365000:365141) kjellander 2015-12-14 21:07:01 -08:00
  • 158879305b Fixing flaky LocalP2PTestSctpDataChannel test. deadbeef 2015-12-14 19:32:34 -08:00
  • c9be00797e Fixing and re-enabling some flaky PeerConnection tests. deadbeef 2015-12-14 18:27:57 -08:00
  • bd292465ee Reland of Free SCTP data channels asynchronously in PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1513143003/ ) deadbeef 2015-12-14 18:15:29 -08:00
  • 82ccfcf5ca Remove unused and rarely used LOG_ macros. solenberg 2015-12-14 08:22:11 -08:00
  • e22e1cb399 Revert of Made EglBase an abstract class and cleaned up. (patchset #4 id:60001 of https://codereview.webrtc.org/1526463002/ ) perkj 2015-12-14 06:43:35 -08:00
  • 40f349fdda [rtp_rtcp] Lint errors cleared from rtp_rtcp/test danilchap 2015-12-14 06:39:33 -08:00
  • 3207916f35 Made EglBase an abstract class and cleaned up. Adds EglBase10 that implemenents EglBase for EGL 1.0 perkj 2015-12-14 06:21:13 -08:00
  • 03960d92dd Roll chromium_revision 4bc4277..10bf0e1 (364953:365000) kjellander 2015-12-14 04:38:11 -08:00
  • bc14164aad Revert of Add APK targets to build libjingle tests for Android. (patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ ) stefan 2015-12-14 04:31:19 -08:00
  • b2f80e3a28 rtp_rtcp/test/BWEStandAlone deleted as obsolete danilchap 2015-12-14 03:21:47 -08:00
  • a78c0211fd Add APK targets to build libjingle_peerconnection_unittests for Android. perkj 2015-12-14 02:41:31 -08:00
  • 17821db197 Wire up bandwidth limitation info to GetStats and adapt_reason. asapersson 2015-12-14 02:08:12 -08:00
  • ac921d7365 Add "x"s in the end of a stripped IPv6 address string. henrikg 2015-12-14 02:07:03 -08:00
  • 38bb8ad1ca Add test for verifying configured key frame interval for VP9. asapersson 2015-12-14 01:41:19 -08:00
  • e5ae6f8237 Correcting the check for the return code produced by AudioProcessing::ProcessReverseStream(). Before the change, only -1 was considered to be an error. peah 2015-12-13 23:02:33 -08:00
  • 1d5c19d23e Address comments from code review 1505253004 (https://codereview.webrtc.org/1505253004/) tommi 2015-12-13 22:54:29 -08:00
  • 4759bfb2a4 Roll chromium_revision 7de03ed..4bc4277 (364770:364953) kjellander 2015-12-13 22:40:46 -08:00
  • aa32c3e537 Update API for Objective-C RTCIceServer hjon 2015-12-13 19:58:11 -08:00
  • cb95f54ee4 Remove pointless move() to fix build on clang/win. Tommi 2015-12-12 16:54:32 +01:00
  • 66679dcc04 Update WARN_UNUSED_RESULT macro to match Chromium's version. tfarina 2015-12-12 05:47:19 -08:00
  • be26c07cb5 Roll gtest-parallel. pbos 2015-12-12 01:45:21 -08:00
  • b798f3883c Roll chromium_revision 710285b..7de03ed (364599:364770) kjellander 2015-12-12 00:24:03 -08:00
  • f888bb58da Support for unmixed remote audio into tracks. Tommi 2015-12-12 01:37:01 +01:00
  • f67c548576 Handle Turn error response to RefreshRequest, CreatePermissionRequest, and ChanelBindRequest Honghai Zhang 2015-12-11 15:16:54 -08:00
  • 04e9146e58 Discard old-generation candidates when ICE restarts The existing code only do so on the controlled side. Honghai Zhang 2015-12-11 14:26:22 -08:00
  • 43e4e23eeb Remove thread-id wraparounds in event tracing. Peter Boström 2015-12-11 20:29:35 +01:00
  • 822bdf9784 Remove cricket::VideoEncoderConfig. Peter Boström 2015-12-11 19:53:39 +01:00
  • 4c1093b86f Add FEC producer fuzzing and a unittest for one of the issues found. Stefan Holmer 2015-12-11 18:25:45 +01:00
  • 5b659c0d10 Special-case android-arm64 in codec bitexactness tests kwiberg 2015-12-11 07:33:59 -08:00
  • b562c33f15 Remove ancient VoE suppressions. solenberg 2015-12-11 05:32:36 -08:00
  • cb23c0d984 Adding Opus to RTPencode. minyue 2015-12-11 01:58:26 -08:00
  • 71f5a9a377 This cl change VideoCaptureAndroid to handle CVO the same way when capturing to texture as when using ordinary byte buffers. Per 2015-12-11 09:32:37 +01:00
  • 0b0a88b918 Add aecdump support to AppRTCDemo aluebs 2015-12-10 23:28:42 -08:00
  • 4dfe332d19 Roll chromium_revision 026b937..710285b (364421:364599) kjellander 2015-12-10 20:22:17 -08:00
  • 55bcf0f087 Fix -Wformat error in Win-Clang build (take 2) hans 2015-12-10 19:49:22 -08:00
  • 013e83b31c Fix -Wformat error in Win-Clang build Niklas Enbom 2015-12-10 16:16:55 -08:00
  • cf846ad60a Adding stub files needed for https://codereview.webrtc.org/1507973003/ Taylor Brandstetter 2015-12-10 15:52:02 -08:00
  • 7c73bdbd82 Renaming JsepPeerConnectionP2PTestClient back to P2PTestConductor. deadbeef 2015-12-10 15:10:44 -08:00
  • ed83edc9e5 Roll chromium_revision 2e451bf..026b937 (364330:364421) kjellander 2015-12-10 13:45:47 -08:00
  • 6a6f0893dd in rtp_rtcp module: fixed build/namespaces lint errors fixed readability/namespace lint errors danilchap 2015-12-10 12:39:08 -08:00
  • a1f567ae90 Revert of Free SCTP data channels asynchronously in PeerConnection. (patchset #3 id:40001 of https://codereview.webrtc.org/1492383002/ ) deadbeef 2015-12-10 11:17:42 -08:00
  • 61a90f94b6 clang/win: Fix -Wextra warnings in webrtc. thakis 2015-12-10 10:50:28 -08:00
  • 5c1def8892 modules/rtp_rtcp/include folder cleared of lint warnings Functions that do not follow lint are marked deprecated, including function in the interface. danilchap 2015-12-10 09:51:54 -08:00
  • 796cfaf7f7 Add VideoCodec::PreferDecodeLate The purpose is so that a decoder (Android) that only have a limited number of output buffers can make sure that decoding is done just before the frame is needed. perkj 2015-12-10 09:27:38 -08:00
  • 4d68208a20 Reduce the runtime of some ACM tests in modules_tests Henrik Lundin 2015-12-10 16:24:39 +01:00
  • c490e01bd1 Implement NativeToI420Buffer in C++, calling java SurfaceTextureHelper, new method .textureToYUV, to do the conversion using an opengl fragment shader. nisse 2015-12-10 06:23:33 -08:00
  • b8b6fbb7a5 lint build/include errors fixed in rtp_rtcp module danilchap 2015-12-10 05:05:27 -08:00
  • 90b9fc9214 Roll chromium_revision a02d286..2e451bf (364268:364330) kjellander 2015-12-10 04:30:30 -08:00
  • 866df6602c Typo fix: Enable a bunch of tests that were accidentally disabled kwiberg 2015-12-10 04:20:02 -08:00
  • 5811a39f14 Replace EventWrapper in video/, test/ and call/. Peter Boström 2015-12-10 13:02:50 +01:00
  • 0f2e939a92 Enable cpplint for more webrtc subfolders and fix all uncovered cpplint errors. jbauch 2015-12-10 03:11:42 -08:00
  • 162abd3562 lint whitespace warning removed from most rtp_rtcp/source/ files rtcp_utility, rtp_utility, tmmbr_help, rtcp_receiver, rtcp_receiver_help are explicetly excluded from the cleanup becaues there are short plans (or cls) to do a deeper cleaning there. danilchap 2015-12-10 02:39:40 -08:00
  • 84e78f9102 Rewrote the PRNG using an xorshift* algorithm and moved the files from test/ to base/. terelius 2015-12-10 01:50:55 -08:00
  • 0b3d7eec07 Prevent RTCP SR to be sent with bogus timestamp. mflodman 2015-12-10 10:10:44 +01:00
  • 48bf2382d9 Some further minor bitexact APM echo suppressor refactoring -Moved memsets to where their variables are used. -Removed redundant. -Changed a pointer scalar to be accessed in pointer notation rather than in array notation. peah 2015-12-09 21:24:53 -08:00
  • 5ba58c6735 Roll chromium_revision dad6346..a02d286 (363782:364268) kjellander 2015-12-09 21:23:03 -08:00
  • a6e4328680 Remove unnecessary test code on Windows. Tommi 2015-12-09 23:27:55 +01:00
  • 70625e5bf3 Enable cpplint for webrtc/examples and fix all uncovered cpplint errors. jbauch 2015-12-09 14:18:14 -08:00
  • 2e5fe31cf6 Remove myself from root_files watchlist. andrew 2015-12-09 12:45:31 -08:00
  • 1387149ad1 Adding reduced size RTCP configuration down to the video stream level. deadbeef 2015-12-09 12:37:51 -08:00
  • ee4082137b WebRTC: Update set of known root certificates Guo-wei Shieh 2015-12-09 11:25:38 -08:00
  • b14f00113e Some minor (bitexact) AEC echo suppressor refactoring -Moved filter reset from the echo suppression into the echo subtraction code where it belongs (the echo subtractor should own its filter reset). -Moved the selection between using the microphone sinal and the echo subtractor output down to the lowest level in the EchoSuppression function. This makes sense as that selection was very hidden in an unrelated sub-sub-function call and as the selection is critical for what the AEC outputs. peah 2015-12-09 11:07:20 -08:00
  • 434aca8d86 Add empty placeholder files for remote audio tracks. This is needed for Chromium so that we can roll, update libjingle.gyp and then continue. tommi 2015-12-09 09:41:54 -08:00
  • afeb43897a Moved code into the lowest level of EchoSuppression to simplify future refactoring and development. peah 2015-12-09 08:50:24 -08:00
  • d1590b2571 Lint clean video/ and add lint presubmit check. mflodman 2015-12-09 07:07:59 -08:00
  • 4cf61dd116 NetEq: Add codec name and RTP timestamp rate to DecoderInfo henrik.lundin 2015-12-09 06:20:58 -08:00
  • 3980d46960 RTCCertificate::Expires() and ::HasExpired() implemented using SSLCertificate::CertificateExpirationTime(). hbos 2015-12-09 05:26:49 -08:00