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  • 4b407aa985 Update AppRTCDemo README with information on 3-dot-apprtc server and new command line arguments. glaznev@webrtc.org 2014-12-04 22:42:59 +00:00
  • 7169afd9d5 With IPv6 enabled, it's important to know whether IPv6 is really used or not. BestConnection is tracked for this purpose. Also added a test case to verify the end to end behavior. guoweis@webrtc.org 2014-12-04 17:59:29 +00:00
  • 369746bcb8 Support new WebSocket signaling format. glaznev@webrtc.org 2014-12-04 17:28:52 +00:00
  • 0b38478885 Add support for parsing header only RTP dumps with bwe_rtp_play. stefan@webrtc.org 2014-12-04 15:43:49 +00:00
  • 9f79fe684a Merge remote bitrate estimator changes. pbos@webrtc.org 2014-12-04 15:34:06 +00:00
  • 33ccdfa1f5 Relanding r7807. minyue@webrtc.org 2014-12-04 12:14:12 +00:00
  • 52bc4f4797 Revert 7807 "Removing unused opus wrapper APIs." minyue@webrtc.org 2014-12-04 11:00:50 +00:00
  • c0991fe606 Roll chromium_revision 24b4c73..f27c369 kjellander@webrtc.org 2014-12-04 10:55:50 +00:00
  • e54a6342dd Removing unused opus wrapper APIs. minyue@webrtc.org 2014-12-04 08:47:25 +00:00
  • 8c9ff203c5 Redo the change of https://webrtc-codereview.appspot.com/30949004/ guoweis@webrtc.org 2014-12-04 07:56:02 +00:00
  • fd8422938c Revert "Implement GetState() for channel's connectivity check state." guoweis@webrtc.org 2014-12-04 00:51:59 +00:00
  • ff72f9e692 Implement GetState() for channel's connectivity check state. guoweis@webrtc.org 2014-12-04 00:14:49 +00:00
  • fd4acf6d55 Adding WebRtcSpl_MaxAbsValueW16 intrinsics version andrew@webrtc.org 2014-12-03 21:59:02 +00:00
  • 3a52458237 add WebRtcIsacfix_AutocorrNeon's intrinsics version andrew@webrtc.org 2014-12-03 21:58:18 +00:00
  • 8dc21dc238 Rename internal AudioEncoder::Encode method to EncodeInternal henrik.lundin@webrtc.org 2014-12-03 20:36:03 +00:00
  • d1fac61e8f Remove need for assembly offset generation in aecm and ns module. andrew@webrtc.org 2014-12-03 17:54:38 +00:00
  • 3800e13a3a Revert r7798 ("Move the AudioDecoder interface out of NetEq") kwiberg@webrtc.org 2014-12-03 16:28:17 +00:00
  • 00ba1a7dfd Move the AudioDecoder interface out of NetEq kwiberg@webrtc.org 2014-12-03 14:23:23 +00:00
  • 0fb6ad2004 Check if cpu_monitor_ exists before Stop(). pbos@webrtc.org 2014-12-03 13:44:29 +00:00
  • fa914e283c Adding a duration printout to neteq_rtpplay henrik.lundin@webrtc.org 2014-12-03 13:28:53 +00:00
  • d8aed6b321 Verify that cpu_monitor exists before calling Stop(). asapersson@webrtc.org 2014-12-03 12:37:47 +00:00
  • c3e097cdc5 Add Android test runner script for WebRTC. kjellander@webrtc.org 2014-12-03 09:57:08 +00:00
  • 8e5c814ef0 Convert DEPS to only reference Git repos kjellander@webrtc.org 2014-12-03 07:11:44 +00:00
  • 511f8a8ef2 TurnPort should ignore STUN binding reponses when using shared socket. jiayl@webrtc.org 2014-12-03 02:17:07 +00:00
  • 001f3b9818 Adjust parameter in videoprocessor_integration_test for vp9. marpan@webrtc.org 2014-12-03 02:00:12 +00:00
  • a7384a1126 Simplify audio_buffer APIs aluebs@webrtc.org 2014-12-03 01:06:35 +00:00
  • ceca014b8b Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeBitRateVP9. marpan@webrtc.org 2014-12-03 01:05:43 +00:00
  • eb0954248d Don't reset sequence number for a stream on deactivate/reactivate. pthatcher@webrtc.org 2014-12-03 00:34:10 +00:00
  • d01955179a Change minimum video encoder initialization resolution to 176x144 to ensure HW encoder can be initialized. glaznev@webrtc.org 2014-12-02 23:41:18 +00:00
  • 1751ee7d32 Remove -flax-vector-conversions flag for ARM NEON building. andrew@webrtc.org 2014-12-02 19:36:14 +00:00
  • ac68ef9ad4 Clear 2 unused functions in audio processing aecm module. andrew@webrtc.org 2014-12-02 18:33:52 +00:00
  • beee9cec22 Change back so that Android ApprtcDemo only use one MediaStream containing both audio and video. The reason is that the desktop apprtcdemo only handle one MediaStream and this doesn't play audio if it receive two streams. perkj@webrtc.org 2014-12-02 14:38:18 +00:00
  • 7f1dfa5b61 Adding a payload type to AudioEncoder objects henrik.lundin@webrtc.org 2014-12-02 12:08:39 +00:00
  • 0cd5558f2b AudioEncoder subclass for G722 kwiberg@webrtc.org 2014-12-02 11:45:51 +00:00
  • 84515f841d Roll chromium_revision 309cf65..24b4c73 kjellander@webrtc.org 2014-12-02 08:48:08 +00:00
  • 5950b645b9 Use c++11 features in webrtc/base/network.cc as a test to see if we can use them. pthatcher@webrtc.org 2014-12-01 23:18:27 +00:00
  • 146e0fd30f Fix the build by putting in a typecast to avoid a comparison between signed and unsigned ints introduced in cl/81073932. pthatcher@webrtc.org 2014-12-01 20:07:52 +00:00
  • dea5173edf Add start bitrate and vp8 hw acceleration option to Android AppRTCDemo. glaznev@webrtc.org 2014-12-01 20:02:13 +00:00
  • 32ec0dd032 (Auto)update libjingle 81063831-> 81073932 buildbot@webrtc.org 2014-12-01 17:57:36 +00:00
  • 7f722492f1 Set simulcastIdx field to zero even if it has no meaning. Helps to be able to memcmp between 2 parses of the same packet. andresp@webrtc.org 2014-12-01 15:29:29 +00:00
  • 273a414b0e Report encoded frame size in VideoSendStream. pbos@webrtc.org 2014-12-01 15:23:21 +00:00
  • 1db20a4180 Adding EncodedInfo struct to AudioEncoder::Encode henrik.lundin@webrtc.org 2014-12-01 14:44:50 +00:00
  • 20446e7e56 Move and rename neteq/test/RTPcat to neteq/tools/rtpcat henrik.lundin@webrtc.org 2014-12-01 14:23:01 +00:00
  • c93437ef96 Add test NetEqDecodingTest.CngFirst henrik.lundin@webrtc.org 2014-12-01 11:42:42 +00:00
  • 83317146ba Adding a new test helper RtpFileWriter and use it in RTPcat henrik.lundin@webrtc.org 2014-12-01 11:25:04 +00:00
  • 4796301c0e Whitespace change to force builds. kjellander@webrtc.org 2014-12-01 09:10:38 +00:00
  • e75f2cea5f Add FORCE_HTTPS_COMMIT_URL to codereview.settings. kjellander@webrtc.org 2014-12-01 09:09:07 +00:00
  • cc7755becd Whitespace change kjellander@webrtc.org 2014-11-29 16:47:53 +00:00
  • 74499efc05 Add whitespace.txt file. kjellander@webrtc.org 2014-11-29 15:42:29 +00:00
  • 2c13f659c7 Add a platform specific typedef for SOCKET in the peerconnection_server example since it's not universally 'int'. tommi@webrtc.org 2014-11-28 10:37:31 +00:00
  • 83b5200f95 Add framerate for complete received frames to histogram stats: "WebRTC.Video.CompleteFramesReceivedPerSecond". asapersson@webrtc.org 2014-11-28 10:17:13 +00:00
  • cc144deaab Make bands vector in SplittingFilter Analysis const aluebs@webrtc.org 2014-11-28 00:26:27 +00:00
  • 8789376cd3 Move ChannelBuffer class to channel_buffer file aluebs@webrtc.org 2014-11-27 23:40:25 +00:00
  • d87213af49 Remove unused RtpStatistics struct. pbos@webrtc.org 2014-11-27 13:48:35 +00:00
  • 7d4e6d012c Roll chromium_revision d8c9041..309cf65 kjellander@webrtc.org 2014-11-27 10:41:04 +00:00
  • d952c40c7e Add receive bitrates to histogram stats: - total bitrate ("WebRTC.Video.BitrateReceivedInKbps") - media bitrate ("WebRTC.Video.MediaBitrateReceivedInKbps") - rtx bitrate ("WebRTC.Video.RtxBitrateReceivedInKbps") - padding bitrate ("WebRTC.Video.PaddingBitrateReceivedInKbps") asapersson@webrtc.org 2014-11-27 07:38:56 +00:00
  • 3e9ad26112 Refactor iOS AppRTC parsing code. tkchin@webrtc.org 2014-11-27 00:52:38 +00:00
  • 79b9eba3ab Implement 3 band splitting filter bank by upsampling and splitting twice into 2 bands aluebs@webrtc.org 2014-11-26 20:21:38 +00:00
  • 7806d8fe40 Fix an ASSERT that fires in a browser test for renegotiation. See https://code.google.com/p/chromium/issues/detail?id=293125#c33 jiayl@webrtc.org 2014-11-26 19:58:50 +00:00
  • a71bb6033b Revert 7750 "Don't reset sequence number for a stream on deactiv..." sprang@webrtc.org 2014-11-26 19:33:15 +00:00
  • a56a2c57cf Enabling building with NEON on ARM64 andrew@webrtc.org 2014-11-26 17:01:40 +00:00
  • 31f7a0e710 Don't reset sequence number for a stream on deactivate/reactivate. sprang@webrtc.org 2014-11-26 16:55:52 +00:00
  • 91d928e737 Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader henrik.lundin@webrtc.org 2014-11-26 15:50:30 +00:00
  • 2faf7eea6f Revert "Revert "This adds an Android apk for running tests on the Java layer of PeerConnection."" perkj@webrtc.org 2014-11-26 07:35:37 +00:00
  • 58edb83fd4 Add video encoder fps and bitrate statistics to Android AppRTCDemo UI. glaznev@webrtc.org 2014-11-26 00:39:42 +00:00
  • 008731868a Implement settable min/start/max bitrates in Call. pbos@webrtc.org 2014-11-25 14:03:34 +00:00
  • b951eb12c9 Add back EXPECT_TRUEs. pbos@webrtc.org 2014-11-25 11:13:28 +00:00
  • ba253473da Reenable GetStats test. pbos@webrtc.org 2014-11-25 09:39:04 +00:00
  • dab5d92df6 Use mirror image for Android AppRTCDemo local preview. glaznev@webrtc.org 2014-11-24 17:31:01 +00:00
  • 03499a0e95 Add wav output capability to neteq_rtpplay henrik.lundin@webrtc.org 2014-11-24 14:50:53 +00:00
  • aff1751c96 Add new test for VP8 packetizer to test tight partitions henrik.lundin@webrtc.org 2014-11-24 12:36:58 +00:00
  • dde19a6f60 sync_chromium.py: Check for chromium/src kjellander@webrtc.org 2014-11-24 10:08:03 +00:00
  • 3398a4ac15 PRESUBMIT: Only notify GN changes for GYP files in webrtc/* kjellander@webrtc.org 2014-11-24 10:05:37 +00:00
  • 8562f23acb OWNERS: Remove tomasl@ and mallinath@ kjellander@webrtc.org 2014-11-24 10:05:05 +00:00
  • 4f16c874c6 Simplifying VideoReceiver and JitterBuffer. pbos@webrtc.org 2014-11-24 09:06:48 +00:00
  • 9334ac2d78 Use vector of CSRCs for DeliverFrame & SetCSRCs. pbos@webrtc.org 2014-11-24 08:25:50 +00:00
  • 308e7ff613 Revert "This adds an Android apk for running tests on the Java layer of PeerConnection." kjellander@webrtc.org 2014-11-23 21:23:00 +00:00
  • 2751f2ab4c This adds an Android apk for running tests on the Java layer of PeerConnection. perkj@webrtc.org 2014-11-23 16:00:57 +00:00
  • 88d14f483b Remove expensive and unnecessary memory alloc for sending black frames on video mute. thorcarpenter@google.com 2014-11-22 01:04:26 +00:00
  • 1153322cf8 Build fix for MIPS Android Webview build. andrew@webrtc.org 2014-11-21 16:28:32 +00:00
  • bdcf38c894 cricket::VideoFrame: Refactor ConvertToRgbBuffer into base class magjed@webrtc.org 2014-11-21 10:53:00 +00:00
  • ad0e71c9a3 Update mock_frame_dropper.h to use size_t kjellander@webrtc.org 2014-11-21 09:40:57 +00:00
  • 4591fbd09f Use size_t more consistently for packet/payload lengths. pkasting@chromium.org 2014-11-20 22:28:14 +00:00
  • edc6e57a92 Support loopback mode and command line execution for Android AppRTCDemo when using WebSocket signaling. glaznev@webrtc.org 2014-11-20 21:16:12 +00:00
  • 6ff3ac1db8 Fix problems if first packet into NetEq is rejected henrik.lundin@webrtc.org 2014-11-20 14:14:49 +00:00
  • ed91068bf1 Create a NetEq test for when the first incoming payload type is unknown henrik.lundin@webrtc.org 2014-11-20 11:01:02 +00:00
  • 049e4ece30 Change default values for CpuOveruseOptions. Enabled method based on encode time and modified values for the low (60->55) and high threshold (90->85). asapersson@webrtc.org 2014-11-20 10:19:46 +00:00
  • f58b455cf7 cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. magjed@webrtc.org 2014-11-19 18:09:14 +00:00
  • 40af3a56e4 Revert "Add DCHECK to ensure that NetEq's packet buffer is not empty" henrik.lundin@webrtc.org 2014-11-19 13:46:52 +00:00
  • 6f6ef72950 Add DCHECK to ensure that NetEq's packet buffer is not empty henrik.lundin@webrtc.org 2014-11-19 13:02:24 +00:00
  • 2176db343c AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. (re-land) henrika@webrtc.org 2014-11-18 13:22:28 +00:00
  • c56814fb2d Roll chromium_revision 91f1781..d8c9041 kjellander@webrtc.org 2014-11-18 10:25:04 +00:00
  • 087da13fe8 Add empty 3 band splitting filter API aluebs@webrtc.org 2014-11-17 23:01:23 +00:00
  • 2656bf813f Fix ExpectedQueueTimeMs() to avoid truncation or overflow. pkasting@chromium.org 2014-11-17 22:21:14 +00:00
  • 930e004a81 Add jmi field for packets discarded due to network error guoweis@webrtc.org 2014-11-17 19:42:14 +00:00
  • c72a22c23d Add preliminary empty file videoframefactory.cc magjed@webrtc.org 2014-11-17 16:34:00 +00:00
  • f5b56fbc41 Annotate COMPILE_ASSERT with __attribute__((unused)). pbos@webrtc.org 2014-11-17 13:47:38 +00:00
  • 4ef22d1d29 Setting Opus FEC as default minyue@webrtc.org 2014-11-17 09:26:39 +00:00
  • 966a708b93 Use RtpFileSource in NetEqDecodingTest henrik.lundin@webrtc.org 2014-11-17 09:08:38 +00:00
  • 4ec19e306a Revert 7707 "cricket::VideoAdapter: Drop frames before spending ..." tommi@webrtc.org 2014-11-16 22:58:11 +00:00