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  • 08df9b2841 Add a manageable command-line tool for AudioProcessing. andrew@webrtc.org 2014-12-16 20:57:15 +00:00
  • cf6d0b64ef Add 48kHz support to AGC aluebs@webrtc.org 2014-12-16 20:56:09 +00:00
  • 2510d11c0f Add (safe) uint32_t cast to fix Win64 build. andrew@webrtc.org 2014-12-16 20:47:42 +00:00
  • 048c5029f5 Handle all permissible PCM fields with WavReader. andrew@webrtc.org 2014-12-16 20:17:21 +00:00
  • 451a133f44 Add AGC manager tests. pbos@webrtc.org 2014-12-16 14:48:47 +00:00
  • c1c9291e9b Make an AudioEncoder subclass for RED henrik.lundin@webrtc.org 2014-12-16 13:41:36 +00:00
  • 88bdec8c3a AudioEncoder subclass for iSACfix kwiberg@webrtc.org 2014-12-16 12:49:37 +00:00
  • 0198933b3d Cleanup: Remove 'const' qualifier from OnReceivedEstimatedBitrate(). kjellander@webrtc.org 2014-12-16 12:29:59 +00:00
  • d08d389ce8 Add field to counters for when first rtp/rtcp packet is sent/received. Use this time for histogram statistics (send/receive bitrates, sent/received rtcp fir/nack packets/min). asapersson@webrtc.org 2014-12-16 12:03:11 +00:00
  • b395a5ea65 audio_processing: Moved legacy AGC code to webrtc/modules/audio_processing/agc/legacy/ bjornv@webrtc.org 2014-12-16 10:38:10 +00:00
  • 55360ae402 Revert "Add adapter_type into Candidate object." guoweis@webrtc.org 2014-12-16 05:28:10 +00:00
  • d021bbbc9e Fix vp9 setting in vie loopback test. marpan@webrtc.org 2014-12-16 00:21:47 +00:00
  • aaf02cc2d4 Add adapter_type into Candidate object. guoweis@webrtc.org 2014-12-15 23:03:10 +00:00
  • 0b1534c52e Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. pkasting@chromium.org 2014-12-15 22:09:40 +00:00
  • 96a626262a Remove 20ms support in AGC aluebs@webrtc.org 2014-12-15 21:54:50 +00:00
  • 1f05c45976 Reenable test case P2PTransportChannelTest.TestIPv6Connections guoweis@webrtc.org 2014-12-15 21:25:54 +00:00
  • a7f77720cb Merge in AGC manager and AGC tools. pbos@webrtc.org 2014-12-15 16:33:16 +00:00
  • 903b4ae603 Removes unused test files by audio_processing/transient bjornv@webrtc.org 2014-12-15 16:13:05 +00:00
  • dd322136fe resources/audio_processing: Removed unused test files bjornv@webrtc.org 2014-12-15 15:57:11 +00:00
  • 6fd9308420 Suppressing warnings in GetRTT() in VoE. minyue@webrtc.org 2014-12-15 14:56:44 +00:00
  • e2e199b894 Clean up StatsObserver's OnComplete methods (address TODOs). tommi@webrtc.org 2014-12-15 13:22:54 +00:00
  • 3440fe1bc5 Use webrtc_root instead of DEPTH for iSAC. pbos@webrtc.org 2014-12-15 10:56:50 +00:00
  • 032b802a8c (Auto)update libjingle 82121498-> 82126219 buildbot@webrtc.org 2014-12-15 09:48:07 +00:00
  • dd0601fbcf Remove unneeded ctor and add a more practical one The default constructor isn't necessary, so I'm removing it. I'm adding another one so that we can (later) make |type| const. tommi@webrtc.org 2014-12-15 09:47:49 +00:00
  • 69bc5a300f Add thread asserts to StatsCollector. Also adding a "ForTest" postfix to a test-only method. tommi@webrtc.org 2014-12-15 09:44:48 +00:00
  • 788acd17ad Merge audio_processing changes. pbos@webrtc.org 2014-12-15 09:41:24 +00:00
  • fb108b5a28 Revert r7885. pbos@webrtc.org 2014-12-15 08:04:50 +00:00
  • b413a30097 Add WebRtcIsacfix_FilterMaLoopNeon's intrinsics version. andrew@webrtc.org 2014-12-15 07:23:49 +00:00
  • 18a3896bd2 Revert r7886:7887. pbos@webrtc.org 2014-12-15 07:03:04 +00:00
  • 40e4767f2b Add NEON intrinsics version for min_max_operations_neon.c andrew@webrtc.org 2014-12-15 06:07:47 +00:00
  • e575e9c40f Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h magjed@webrtc.org 2014-12-14 11:09:23 +00:00
  • e9db7fe80c Put pseudotcp back because remoting uses it. pthatcher@webrtc.org 2014-12-13 01:56:39 +00:00
  • dee76f3b89 Move the obvious/easy Jingle-specific code into webrtc/libjingle. pthatcher@webrtc.org 2014-12-12 21:04:42 +00:00
  • 8c9d79a29d Add adapter_type into Candidate object. guoweis@webrtc.org 2014-12-12 19:21:14 +00:00
  • c57310b982 Switch kStatsValueName* constants to be enums instead of char*. This is to guard against potentially assigning a value name to an incorrect value, non-static string or otherwise assume they can be treated as strings. tommi@webrtc.org 2014-12-12 17:41:28 +00:00
  • 3b79daff14 Moving encoded_bytes into EncodedInfo henrik.lundin@webrtc.org 2014-12-12 13:31:24 +00:00
  • c8bc717905 Fix webrtc gn windows build. kjellander@webrtc.org 2014-12-12 12:10:46 +00:00
  • f68faa542a Removing manual test pages because they have been moved to github. jansson@webrtc.org 2014-12-12 09:30:41 +00:00
  • 40b276ea7b Cleanup little things found when refactoring. pthatcher@webrtc.org 2014-12-12 02:44:30 +00:00
  • 27d106bcf7 Move the downmixing out of AudioBuffer aluebs@webrtc.org 2014-12-11 17:09:21 +00:00
  • 0ca768b131 Adding DTX to WebRTC Opus wrapper (relanding). minyue@webrtc.org 2014-12-11 16:09:35 +00:00
  • 5f162c8509 Merge AEC changes. pbos@webrtc.org 2014-12-11 13:46:59 +00:00
  • 2b19f06312 Wire up RTT statistics to webrtc::Call. pbos@webrtc.org 2014-12-11 13:26:09 +00:00
  • 13518951e3 Remove old_factory from WebRtcVideoEngine. pbos@webrtc.org 2014-12-11 13:14:30 +00:00
  • 128fabaf7b Revert "Revert 7826 "Change Android PeerConnectionUnittest to build usin..."" perkj@webrtc.org 2014-12-11 12:25:57 +00:00
  • 626c09f6a3 Move isolate path into webrtc/build/android/test_runner.py kjellander@webrtc.org 2014-12-11 11:59:46 +00:00
  • 817e50dd7d Make an AudioEncoder subclass for PCM16B henrik.lundin@webrtc.org 2014-12-11 10:47:19 +00:00
  • b3ad8cf6ca Make an AudioEncoder subclass for iSAC kwiberg@webrtc.org 2014-12-11 10:08:19 +00:00
  • abe3f1879c Checking whether ACM uses codec internal or WebRTC DTX. minyue@webrtc.org 2014-12-11 08:53:21 +00:00
  • 55d42c32a4 DCHECK: Reference condition parameter in release builds kwiberg@webrtc.org 2014-12-11 08:32:30 +00:00
  • cd5b209d68 Deleting quality dashboard code. phoglund@webrtc.org 2014-12-11 07:57:22 +00:00
  • 3c31e6e2f9 Add NEON intrinsics version for WebRtcSpl_MinValueW16Neon andrew@webrtc.org 2014-12-11 00:24:13 +00:00
  • f4c19480fc Remove jitter_estimate_test.h mflodman@webrtc.org 2014-12-10 21:08:39 +00:00
  • c5ebbd98f5 Support 48kHz in Noise Suppression aluebs@webrtc.org 2014-12-10 19:30:57 +00:00
  • d8ca723de7 Remove CELT support from audio_coding. pbos@webrtc.org 2014-12-10 11:49:13 +00:00
  • 8084f9500f Change LastProcessedRtt (used in the rtp/rtcp module) to return the average RTT (instead of max RTT) to get a smooth estimate of the nack interval. asapersson@webrtc.org 2014-12-10 11:04:13 +00:00
  • 85bd53e7c9 Add AbsSendTime unittests to rampup_tests.cc. pbos@webrtc.org 2014-12-10 10:36:20 +00:00
  • 0df371549f Cast payload type to int in logs. asapersson@webrtc.org 2014-12-10 10:30:45 +00:00
  • a85307737c (Auto)update libjingle 81702493-> 81755413 buildbot@webrtc.org 2014-12-10 09:01:18 +00:00
  • 3cd26b677a Revert r7858 ("DCHECK: Reference condition parameter in release builds") kwiberg@webrtc.org 2014-12-10 08:57:14 +00:00
  • 3148060e61 DCHECK: Reference condition parameter in release builds kwiberg@webrtc.org 2014-12-10 08:45:47 +00:00
  • ff1a3e36bd Make an AudioEncoder subclass for comfort noise henrik.lundin@webrtc.org 2014-12-10 07:29:08 +00:00
  • 6fd52f36db Add NEON intrinsics version for WebRtcSpl_DownsampleFastNeon. andrew@webrtc.org 2014-12-10 00:59:48 +00:00
  • ae20d3bbce Add NEON intrinsics version for WebRtcSpl_CrossCorrelationNeon. andrew@webrtc.org 2014-12-09 23:58:39 +00:00
  • aa2c342c10 Add back a constructor to fix FYI build. tommi@webrtc.org 2014-12-09 20:23:06 +00:00
  • 5c32a84620 Attempt to fix FYI bots. The FYI bots went red after https://webrtc-codereview.appspot.com/32179004/ landed. tommi@webrtc.org 2014-12-09 19:59:27 +00:00
  • 87776a8935 iAppRTCDemo: WebSocket based signaling. tkchin@webrtc.org 2014-12-09 19:32:35 +00:00
  • 0babb4a4e6 Fix a comment. pthatcher@webrtc.org 2014-12-09 19:01:45 +00:00
  • c9d155faeb Move implementation of types in statstypes. to its cc file. tommi@webrtc.org 2014-12-09 18:18:06 +00:00
  • a954c07ee1 AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer henrika@webrtc.org 2014-12-09 16:22:09 +00:00
  • 19dd129c69 Revert 7846 "Adding DTX to WebRTC Opus wrapper" minyue@webrtc.org 2014-12-09 15:11:15 +00:00
  • f244760827 Add histograms for receive statistics: - decoded frames per second ("WebRTC.Video.DecodedFramesPerSecond") - percentage of delayed frames to rendered ("WebRTC.Video.DelayedFramesToRenderer") - average delay (of delayed frames) to renderer ("WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs") asapersson@webrtc.org 2014-12-09 14:13:26 +00:00
  • 4321f175f1 Adding DTX to WebRTC Opus wrapper minyue@webrtc.org 2014-12-09 13:27:39 +00:00
  • 5c3ee4bce6 Add empty implementation file that will hold statstypes.h implementation. The implementation for the types currently in statstypes.h is split between statstypes.h and statscollector.cc. tommi@webrtc.org 2014-12-09 10:47:01 +00:00
  • 1784d7cfad Adding an codec interal CNG test in NetEq. minyue@webrtc.org 2014-12-09 10:46:39 +00:00
  • 9115cde6c9 Merge VP8 changes. pbos@webrtc.org 2014-12-09 10:36:40 +00:00
  • e04a93bcf5 Move the AudioDecoder interface out of NetEq kwiberg@webrtc.org 2014-12-09 10:12:53 +00:00
  • 97d0489058 Add video send bitrates to histogram stats: - total bitrate ("WebRTC.Video.BitrateSentInKbps") - media bitrate ("WebRTC.Video.MediaBitrateSentInKbps") - rtx bitrate ("WebRTC.Video.RtxBitrateSentInKbps") - padding bitrate ("WebRTC.Video.PaddingBitrateSentInKbps") - retransmitted bitrate ("WebRTC.Video.RetransmittedBitrateInKbps") asapersson@webrtc.org 2014-12-09 09:47:53 +00:00
  • 7ba9f27f2b Set CHECKOUT_SOURCE_ROOT environment variable for Android test wrapper. kjellander@webrtc.org 2014-12-09 06:46:13 +00:00
  • eef85387ec Fix AppRTCDemo closing error for KK and JB Android devices. glaznev@webrtc.org 2014-12-09 01:29:17 +00:00
  • 86b6d65ef1 Remove no longer used video codec test framework. stefan@webrtc.org 2014-12-09 00:02:45 +00:00
  • 8911bc52f1 Add AudioEncoder::Max10MsFramesInAPacket henrik.lundin@webrtc.org 2014-12-08 21:15:55 +00:00
  • 130fef89dd Bugfix in AudioDecoderTest henrik.lundin@webrtc.org 2014-12-08 21:07:59 +00:00
  • edeea91803 Change all system clock types to int64_t in bitrate_controller. stefan@webrtc.org 2014-12-08 19:46:23 +00:00
  • fcbe36a1d9 Add const qualifier to WebRtcPcm16b_Encode henrik.lundin@webrtc.org 2014-12-08 18:26:49 +00:00
  • a1ef7bfa15 ATTRIBUTE_UNUSED expanded to empty on MSVS, so be sure to use the variable. kwiberg@webrtc.org 2014-12-08 17:53:10 +00:00
  • 3b3c406908 Revert 7826 "Change Android PeerConnectionUnittest to build usin..." andrew@webrtc.org 2014-12-08 17:21:50 +00:00
  • cb858ba397 Make an AudioEncoder subclass for iLBC kwiberg@webrtc.org 2014-12-08 17:11:44 +00:00
  • ee43263a50 Cleaned up real_fft APIs due to non-existing NEON code bjornv@webrtc.org 2014-12-08 16:36:22 +00:00
  • ed7824b1c0 Change Android PeerConnectionUnittest to build using Chrome macros. The purpose is to be able to run the tests using Chromes buildbots. To run: CHECKOUT_SOURCE_ROOT=pwd build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest perkj@webrtc.org 2014-12-08 15:41:01 +00:00
  • ba8138ba38 Change type of nack_last_time_sent_full_ from uint32_t to int64_t. Could cause nack requests to be sent too frequently. asapersson@webrtc.org 2014-12-08 13:29:02 +00:00
  • aefe61ae2a PRESUBMIT: Add check for checkdeps. kjellander@webrtc.org 2014-12-08 13:00:30 +00:00
  • 7db359b94a Roll chromium_revision 24b4c73..8e72e1d kjellander@webrtc.org 2014-12-08 11:48:35 +00:00
  • d91d359feb PRESUBMIT: Add iOS ARM64 trybots to default set. kjellander@webrtc.org 2014-12-08 07:05:38 +00:00
  • fb01376eca Adjust some parameters for VP9 tests. marpan@webrtc.org 2014-12-08 06:25:51 +00:00
  • e2a9261f3e Improve AppRTCDemo connection speed by sending all http POST requests asynchronously. glaznev@webrtc.org 2014-12-05 20:11:06 +00:00
  • bd8cc0b914 Add codereview.settings to the /talk subdirectory kjellander@webrtc.org 2014-12-05 13:47:37 +00:00
  • 5af8cd77e2 Add codereview.settings to the /webrtc subdirectory kjellander@webrtc.org 2014-12-05 13:43:35 +00:00
  • 599e299b9d cricket::VideoFrame int64 to int64_t. kjellander@webrtc.org 2014-12-05 09:42:57 +00:00
  • 9b5467e88d Fix assertion failure when closing data channel, and add a unit test. bemasc@webrtc.org 2014-12-04 23:16:52 +00:00