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  • af2f3dd206 Reland: Add IntelligibilityEnhancer support to audioproc_float Alejandro Luebs 2016-03-29 14:54:37 -07:00
  • d45b95c270 Making new unit test assertions use the standard timeout. Taylor Brandstetter 2016-03-29 13:16:52 -07:00
  • dd56fa8642 Revert "Add IntelligibilityEnhancer support to audioproc_float" Alejandro Luebs 2016-03-29 13:05:40 -07:00
  • 98c69a0ee7 Add IntelligibilityEnhancer support to audioproc_float Alejandro Luebs 2016-03-29 12:43:32 -07:00
  • f537768497 Update QuicTransportChannel to latest version of libquic mikescarlett 2016-03-29 12:14:55 -07:00
  • 9708e9c599 Don't call operator== with scoped_ptr<T> and T* kwiberg 2016-03-29 10:17:42 -07:00
  • 345807e599 Remove calls to rtc::scoped_ptr::accept kwiberg 2016-03-29 10:16:34 -07:00
  • 57d5a2e4df Reland of Added a bitexactness test for the gain controller in the audio processing module. peah 2016-03-29 09:48:36 -07:00
  • 4079cc301b Remove accidentally readded webrtc/base/sslstreamadapterhelper.cc Henrik Kjellander 2016-03-29 17:52:21 +02:00
  • b252856d10 Remove all uses of the HAVE_CONFIG_H define. Henrik Kjellander 2016-03-29 17:47:19 +02:00
  • 2f36c2399e [rtcp] SenderReport::Parse updated not to use RTCPUtility danilchap 2016-03-29 08:02:27 -07:00
  • f81603549a Move to x509 v3 as required by the WebRTC draft. torbjorng 2016-03-29 07:57:43 -07:00
  • c02c0a7fd2 Remove orphaned files. torbjorng 2016-03-29 07:22:09 -07:00
  • ceef04613b Added a bitexactness test for the beamformer in the audio processing module peah 2016-03-29 05:35:48 -07:00
  • 2378212e5a Android HW decoder: Add support for textures when using EGL 1.0 magjed 2016-03-29 05:13:21 -07:00
  • 9c246c4bbe Change include in metrics.h (change to use systems_wrappers/include/logging.h, base logging breaks chromium.fyi). asapersson 2016-03-29 04:16:24 -07:00
  • ae69b02846 Fix typo in FakeAdmTest.TestProcess name. Peter Boström 2016-03-29 12:12:57 +02:00
  • 58d992e025 Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*). asapersson 2016-03-29 02:15:06 -07:00
  • 6b1968ea70 Allow passing in strings of length zero to FileWrapper::Write without closing the file. terelius 2016-03-29 01:39:59 -07:00
  • d4f6ea70b5 Re-reland of Added a bitexactness test for the echo canceller in the audio processing module. peah 2016-03-29 00:37:44 -07:00
  • 85e46a8f13 Fix PeerConnectionInterfaceTest.CloseAndTestStreamsAndStates Increase delay for waiting for AudioSource state change to 10ms to hopefully fix flake that started happening on the 24th of Mars after landing https://codereview.webrtc.org/1816143002/. Per 2016-03-29 09:18:05 +02:00
  • 4b0c74172e Added a bitexactness test for the intelligibility enhancer in the audio processing module peah 2016-03-28 23:43:49 -07:00
  • 3a2f7e01ef Build dynamic framework with podspec for Objective-C API. Jon Hjelle 2016-03-25 14:48:14 -07:00
  • 918d015737 Revert of Added a bitexactness test for the gain controller in the audio processing module. (patchset #3 id:60001 of https://codereview.webrtc.org/1812433002/ ) peah 2016-03-25 00:50:42 -07:00
  • 3587e331a8 Roll chromium_revision c656a0e..355fabf (383013:383228) kjellander 2016-03-24 21:01:09 -07:00
  • 303b3c21a4 Added the JNI interface to get and set RtpParameters and the maximum bitrate limits. Defined a JavaCollection convenience class to simplify iterating over collections from within JNI code Follow-up to https://codereview.webrtc.org/1788583004/. BUG= skvlad 2016-03-24 19:36:46 -07:00
  • a49dc36976 Added a bitexactness test for the gain controller in the audio processing module. peah 2016-03-24 17:59:20 -07:00
  • 84db6fa7f5 Adding BlockMeanCalculator for AEC. minyue 2016-03-24 14:36:25 -07:00
  • 7c931ad698 Fixed a potential deadlock problem in the AGC where the render and capture locks were acquired in the wrong order. peah 2016-03-24 12:52:02 -07:00
  • 1c2af8e319 Avoid clicks when muting/unmuting a voe::Channel. solenberg 2016-03-24 10:36:00 -07:00
  • 60d5f3f4b7 Don't override curve preferences in BoringSSL. David Benjamin 2016-03-24 13:28:25 -04:00
  • 027fd8f907 Revert of Added a bitexactness test for the echo canceller in the audio processing module. (patchset #2 id:40001 of https://codereview.webrtc.org/1827833006/ ) guidou 2016-03-24 06:43:59 -07:00
  • 7ca142e047 ReAdd dummy MediaStreamTrack::set_state to make Chrome build happy. perkj 2016-03-24 14:19:08 +01:00
  • 9adc91d9cc Revert of Remove code interfacing legacy openssl. (patchset #3 id:40001 of https://codereview.webrtc.org/1808763002/ ) Torbjorn Granlund 2016-03-24 14:05:06 +01:00
  • 4a3a135e68 More cleanup of cricket::VideoCapturer Remove unused state CS_PAUSED and dummy implementation of Pause() Remove unused SignalVideoFrame perkj 2016-03-24 04:39:49 -07:00
  • d61bf803d2 Removed MediaStreamTrackInterface::set_state The track state should be implicitly set by the underlying source. This removes the public method and cleans up how AudioRtpReceiver is created. Further more it cleans up how the RtpReceivers are destroyed. perkj 2016-03-24 03:16:19 -07:00
  • e29f0e2515 Reland of Added a bitexactness test for the echo canceller in the audio processing module. peah 2016-03-24 02:45:54 -07:00
  • 85fa7d5311 Move swap_queue.h to base/ terelius 2016-03-24 01:51:52 -07:00
  • a4f07887c7 Delete default_send_ssrc_. nisse 2016-03-24 01:02:49 -07:00
  • 75a2c2386c Roll chromium_revision 004c6f1..c656a0e (382886:383013) kjellander 2016-03-23 20:25:20 -07:00
  • a0c44eaa82 Add 16-bit network id to the candidate signaling. Also include that in the stun-ping request as part of the network-info attribute. Change the network cost to be 16 bits. honghaiz 2016-03-23 16:07:48 -07:00
  • 887a19b9d2 Switch to using EGL 1.0 for rendering and HW codec. Alex Glaznev 2016-03-23 15:01:44 -07:00
  • 1bd9553d2e Add visibility flag to GYP. tkchin 2016-03-23 13:19:18 -07:00
  • 038cea3968 Roll chromium_revision 6d73369..004c6f1 (382825:382886) kjellander 2016-03-23 13:01:04 -07:00
  • 24a62d5d83 Remove WEBRTC_IOS from RTCPeerConnectionFactory public header. tkchin 2016-03-23 11:29:27 -07:00
  • a8415fe9ea Adding comments about threading around CreatePeerConnectionFactory. Taylor Brandstetter 2016-03-23 10:38:07 -07:00
  • f752f85f3d [rtcp] Pli::Parse updated not to use RTCPUtility danilchap 2016-03-23 08:25:25 -07:00
  • 2f294b4092 Roll chromium_revision 3f0d1a9..6d73369 (382768:382825) kjellander 2016-03-23 06:04:08 -07:00
  • 7ade7b3282 Delete class webrtc::VideoRenderer and its header file. nisse 2016-03-23 04:48:10 -07:00
  • 1509fa1aa9 Delete cricket::VideoRenderer. nisse 2016-03-23 04:05:57 -07:00
  • de3185521b Add Mic Toggle button to AppRTCDemo (Android). solenberg 2016-03-23 02:57:06 -07:00
  • 8f59762897 Delete VideoRendererInterface. Niels Möller 2016-03-23 10:33:07 +01:00
  • c8f952deaa Propagate MediaStreamSource state to video tracks the same way as audio. Also removes unused track states kLive and kFailed. Since this also required a Video source to exist in all unit tests that create a track, a FakeVideoTrackSource is added and used in tests. perkj 2016-03-23 00:33:56 -07:00
  • 8549ed7638 Roll chromium_revision 029c2f4..3f0d1a9 (382621:382768) kjellander 2016-03-22 22:27:29 -07:00
  • 752f36fc2a Roll chromium_revision ba603a0..029c2f4 (381748:382621) kjellander 2016-03-22 16:56:01 -07:00
  • 2df29cb673 Remove redefined macros from BitrateAdjuster emircan 2016-03-22 16:05:27 -07:00
  • dbe2b8744f Adding support for RTCRtpEncodingParameters.active flag. deadbeef 2016-03-22 15:42:00 -07:00
  • 7a43d253f9 Make the audio channel communicate network state changes to the call. skvlad 2016-03-22 15:32:27 -07:00
  • 01bcbd0df6 Make Android min-resolution rotation-agnostic. Peter Boström 2016-03-22 21:44:29 +01:00
  • 56036ffc45 cleanup RTCPSender rtc::scoped_ptr -> std::unique_ptr. CrticalSectionWrapper -> CriticalSection. assert -> DCHECK. removed unused headers. removed unused using. removed unused member field. danilchap 2016-03-22 11:14:09 -07:00
  • 8d2ade65b1 Reland of Added a bitexactness test for the echo control mobile in the audio processing module peah 2016-03-22 11:05:12 -07:00
  • f3cb49f3ef Refactor some ObjC API init methods. Tze Kwang Chin 2016-03-22 10:57:40 -07:00
  • 09c3a1e291 Use rtc::scoped_refptr for WebRtcVideoCapturer. Peter Boström 2016-03-22 17:17:39 +01:00
  • 2943f015b6 Reland of VCMCodecTimer: Change filter from max to 95th percentile (patchset #1 id:1 of https://codereview.webrtc.org/1808693002/ ) magjed 2016-03-22 05:12:09 -07:00
  • 81cbd92444 Revert of Initialize/configure video encoders asychronously. (patchset #4 id:60001 of https://codereview.webrtc.org/1757313002/ ) Peter Boström 2016-03-22 12:19:07 +01:00
  • 60624cd6bf [iOS] Link with base/maccocoathreadhelper.mm on iOS. sdefresne 2016-03-22 03:32:14 -07:00
  • 6a85d3450c Fixing UpdateLevel function in AEC. minyue 2016-03-22 02:14:53 -07:00
  • 2cb73413f4 Moved sequence number specific operations from mod_ops.h to sequence_number_util.h philipel 2016-03-22 10:03:43 +01:00
  • 53cf3463c0 Fix race condition in EventTimerPosix sprang 2016-03-22 01:51:39 -07:00
  • e56b99ed02 Update CPU Monitor to report CPU frequency and battery level. Alex Glaznev 2016-03-21 16:24:39 -07:00
  • 307a0922c5 Support delayed AudioUnit initialization. Tze Kwang Chin 2016-03-21 13:57:40 -07:00
  • bc73fe1aad Move build scripts to webrtc/build/ios hjon 2016-03-21 11:38:26 -07:00
  • 0db3db94e5 Put config in sync between gyp and gn. sdefresne 2016-03-21 11:20:28 -07:00
  • 0c4de568a6 Fix potential crashes in the screen capturer on Mac Sergey Ulanov 2016-03-21 11:18:42 -07:00
  • 121ac122b9 Fix some warnings in ObjC code. tkchin 2016-03-21 09:08:40 -07:00
  • 1d1944187f Replace RefCountImpl with rtc::RefCountedObject. Peter Boström 2016-03-21 16:44:31 +01:00
  • af510afc91 Use a FakeVideoTrackSource instead of nullptr in all VideoTrack tests. nisse 2016-03-21 08:20:42 -07:00
  • c5dabdd3fb Add support for configuring the number of spatial/temporal layers for VP9 through a field trial. asapersson 2016-03-21 04:15:50 -07:00
  • f26f98b29c Revert of Added a bitexactness test for the echo canceller in the audio processing module. (patchset #3 id:60001 of https://codereview.webrtc.org/1809613002/ ) peah 2016-03-21 02:35:18 -07:00
  • b60be20957 Revert of Added a bitexactness test for the echo control mobile in the audio processing module (patchset #3 id:60001 of https://codereview.webrtc.org/1805373002/ ) peah 2016-03-21 02:34:10 -07:00
  • eb83a1a10f This is an initial cleanup step, aiming to delete the webrtc::VideoRenderer class, replacing it by rtc::VideoSinkInterface. nisse 2016-03-21 01:27:56 -07:00
  • 105831ef4a Added a bitexactness test for the echo control mobile in the audio processing module peah 2016-03-21 01:10:20 -07:00
  • 7c448e1a38 Added a bitexactness test for the echo canceller in the audio processing module. peah 2016-03-20 17:22:20 -07:00
  • 62411a21c9 Fixing crash that may occur after destroying a VideoSendStream. deadbeef 2016-03-20 14:24:49 -07:00
  • bdbceeffe8 Added a bitexactness test for the voice activity detector in the audio processing module. peah 2016-03-20 09:53:32 -07:00
  • 9e083d2ac5 Reland of Delete empty API files and cleaned up includes. (patchset #1 id:1 of https://codereview.webrtc.org/1813083002/ ) perkj 2016-03-20 09:38:40 -07:00
  • 19b7b665cc Added a bitexactness test for the level estimator in the audio processing module. peah 2016-03-20 08:36:28 -07:00
  • caafdba0e4 Fix broken CVO header extension perkj 2016-03-20 07:34:29 -07:00
  • eec21bdae3 Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. jbauch 2016-03-20 06:15:43 -07:00
  • 5585001e5d Added a bitexactness test for the noise suppressor. peah 2016-03-19 18:01:09 -07:00
  • 194e3bcc53 Revert of Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. (patchset #4 id:60001 of https://codereview.webrtc.org/1785713005/ ) kjellander 2016-03-19 12:12:52 -07:00
  • ebfbab5059 Move copyonwritebuffer to rtc_base_approved. The other buffer classes as well as all other dependencies are in rtc_base_approved, so I think this is a better place for it. Additionally I found that code in Chromium that already depends on the other buffer classes but now depends on the CopyOnWriteBuffer class, needed to have their build files updated and they previously depended on the buffer classes in rtc_base_approved. tommi 2016-03-19 11:36:18 -07:00
  • 944c39006f Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. jbauch 2016-03-19 01:57:31 -07:00
  • 5f0b83b7fb Enabling rtcp-rsize negotiation and fixing some issues with it. Taylor Brandstetter 2016-03-18 15:02:07 -07:00
  • 1300caa3fe Refactor AudioUnit code into its own class. Zeke Chin 2016-03-18 14:39:11 -07:00
  • 433b95a685 Fixing issues with timestamps in video_quality_test.cc. Taylor Brandstetter 2016-03-18 11:41:03 -07:00
  • f5629ad44f Reland of Change VideoCapture_apply_rotation default to false (patchset #1 id:1 of https://codereview.webrtc.org/1807673008/ ) deadbeef 2016-03-18 11:38:26 -07:00
  • 223b982785 Revert of Change VideoCapture_apply_rotation default to false (patchset #4 id:80001 of https://codereview.webrtc.org/1779883004/ ) perkj 2016-03-18 10:06:29 -07:00
  • 6919457319 Change VideoCapture_apply_rotation default to false Per 2016-03-18 16:56:47 +01:00
  • 3b4d074452 Remove webrtc/sound/DEPS. solenberg 2016-03-18 06:25:24 -07:00