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  • d40e404be4 Revert r3815 marpan@webrtc.org 2013-04-10 21:37:03 +00:00
  • 1b2a6e0be0 Updated WebRTC version number to 3.29 TBR=mallinath1 Review URL: https://webrtc-codereview.appspot.com/1305005 elham@webrtc.org 2013-04-10 21:28:33 +00:00
  • 6f41ca9fd2 WebRTCDemo: Enable making multiple calls. Previously after the first call subsequent attempts to bind the RTP/RTCP ports would fail, since r3754. fischman@webrtc.org 2013-04-10 20:33:27 +00:00
  • 59d8889704 Add OWNERS file for channel_transport kjellander@webrtc.org 2013-04-10 20:16:26 +00:00
  • 6bfcbcda13 Roll libvpx to 192165. -pick up libvpx roll to 3db60c8. marpan@webrtc.org 2013-04-10 20:08:53 +00:00
  • e4b6064f8e Replace legacy G_CONST with const. pbos@webrtc.org 2013-04-10 18:06:57 +00:00
  • ab9202b673 Removing remaining WebRtc_Word32 not in typedefs.h pbos@webrtc.org 2013-04-10 17:59:17 +00:00
  • 77d59fe408 WebRTCDemo: no-op out instead of NPEing on destroyed camera. fischman@webrtc.org 2013-04-10 17:11:51 +00:00
  • dfc5bb9c97 WebRtc_Word32 -> int32_t in video_capture/ pbos@webrtc.org 2013-04-10 08:23:13 +00:00
  • ddf94e71e5 WebRtc_Word32 -> int32_t in video_render/ pbos@webrtc.org 2013-04-10 08:09:04 +00:00
  • b7192b8247 WebRtc_Word32 -> int32_t in audio_processing/ pbos@webrtc.org 2013-04-10 07:50:54 +00:00
  • 557e92515d Reapply the reverted r3747. marpan@webrtc.org 2013-04-09 21:21:32 +00:00
  • 806dc3b0e6 More trace events hclam@chromium.org 2013-04-09 19:54:10 +00:00
  • 4d2f5de67a Improve how NACK lists are generated before a frame has been decoded. stefan@webrtc.org 2013-04-09 18:24:41 +00:00
  • ac891627c6 WebRtc_Word32 -> int32_t in audio_conference_mixer/ pbos@webrtc.org 2013-04-09 17:40:15 +00:00
  • b09130763b WebRtc_Word32 -> int32_t in common_audio/ pbos@webrtc.org 2013-04-09 16:40:28 +00:00
  • 7da3459b2a Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps." stefan@webrtc.org 2013-04-09 14:56:29 +00:00
  • b238d1210b WebRtc_Word32 -> int32_t in video_engine/ pbos@webrtc.org 2013-04-09 13:41:51 +00:00
  • 1ab45f6dd5 WebRtc_Word32 -> int32_t in video_processing/ pbos@webrtc.org 2013-04-09 13:38:10 +00:00
  • afcc6101d0 With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps. stefan@webrtc.org 2013-04-09 13:37:40 +00:00
  • fd2bfc8fca WebRtc_Word32 -> int32_t in common_video. pbos@webrtc.org 2013-04-09 13:36:08 +00:00
  • c75102eba7 WebRtc_Word32 -> int32_t in utility/ pbos@webrtc.org 2013-04-09 13:32:55 +00:00
  • 0ea11c1768 WebRtc_Word32 -> int32_t in media_file/ pbos@webrtc.org 2013-04-09 13:31:37 +00:00
  • a701c0ed03 Fixing the flakiness of ThreadWakesTwice. hta@webrtc.org 2013-04-09 12:36:28 +00:00
  • a5f1787f63 WebRtc_Word32 -> int32_t in test/ pbos@webrtc.org 2013-04-09 11:10:21 +00:00
  • 2550988baa WebRtc_Word32 -> int32_t in audio_device/ pbos@webrtc.org 2013-04-09 10:30:35 +00:00
  • 6141e13873 WebRtc_Word32 -> int32_t in voice_engine/ pbos@webrtc.org 2013-04-09 10:09:10 +00:00
  • 046deb9b20 WebRtc_Word32 -> int32_t in system_wrappers pbos@webrtc.org 2013-04-09 09:06:11 +00:00
  • 29758de9b6 Always set render delay in ViEChannel::RegisterExternalDecoder. pbos@webrtc.org 2013-04-09 00:34:42 +00:00
  • 0946a56023 WebRtc_Word32 => int32_t etc. in audio_coding/ pbos@webrtc.org 2013-04-09 00:28:06 +00:00
  • 6faf71d27b Remove the old unused udp_transport Review URL: https://webrtc-codereview.appspot.com/1272009 pwestin@webrtc.org 2013-04-08 23:25:25 +00:00
  • 4c44fe0561 Updated pranswer, dtmf demos & deleted pc1-deprecated.html. Review URL: https://webrtc-codereview.appspot.com/1287007 vikasmarwaha@webrtc.org 2013-04-08 21:23:58 +00:00
  • 6ff76c7404 Reduce execution time of rate control test. marpan@webrtc.org 2013-04-08 20:32:48 +00:00
  • cf8e108158 Fixed a bug in isac-fix's entropy coding function: out of bounds acces to array. BUG=227286 Review URL: https://webrtc-codereview.appspot.com/1293005 kma@webrtc.org 2013-04-08 16:37:53 +00:00
  • b4a0623e43 Fix of lint script errors in apprtc.py pbos@webrtc.org 2013-04-08 15:59:24 +00:00
  • f2e7bc6b6a Added maxlen=80 to CheckLongLines() call in PRESUBMIT.py pbos@webrtc.org 2013-04-08 15:46:07 +00:00
  • 034f004a4f WebRtc_Word32 => int32_t in video_coding/ pbos@webrtc.org 2013-04-08 11:13:29 +00:00
  • 2f44673d66 WebRtc_Word32 => int32_t for rtp_rtcp/ pbos@webrtc.org 2013-04-08 11:08:41 +00:00
  • 367804cce2 Clean packets on the network when closing + made loopback test actually run again. mflodman@webrtc.org 2013-04-08 10:42:50 +00:00
  • ff7e1303e8 WebRtc_Word32 => int32_t remote_bitrate_estimator/ pbos@webrtc.org 2013-04-08 10:04:37 +00:00
  • 37bf5847dc Show stats from both sides hta@webrtc.org 2013-04-06 10:05:55 +00:00
  • 222e9948f5 Migrating Apprtc to use new TURN service which supports time-limited TURN credentials. Review URL: https://webrtc-codereview.appspot.com/1291004 vikasmarwaha@webrtc.org 2013-04-06 05:58:15 +00:00
  • 123b618f48 Fix a crash issue on WinXP where LoadLibrary(TEXT("Kernel32.dll")) may fail. wu@webrtc.org 2013-04-06 04:13:51 +00:00
  • 2e6b7e938f In streaming mode it is preferable to fade to silence when sender stops sending, or long period of packet loss. turaj@webrtc.org 2013-04-06 00:08:11 +00:00
  • 19da719a5f Resolves TSan v2 reports data races in voe_auto_test. henrika@webrtc.org 2013-04-05 14:34:57 +00:00
  • 10eb92039b Add GYP target for WebRTC Video demo for Android. kjellander@webrtc.org 2013-04-05 13:36:32 +00:00
  • b5bf54c4e7 Permit arbitrary payload names for kVideoCodecGeneric. pbos@webrtc.org 2013-04-05 13:27:38 +00:00
  • b9e402d99f Remove WEBRTC_*_ENGINE_NETWORK_API use Review URL: https://webrtc-codereview.appspot.com/1203009 pwestin@webrtc.org 2013-04-04 19:51:42 +00:00
  • 79b0289bfc Adds event traces and counters for WebRTC receive side. Review URL: https://webrtc-codereview.appspot.com/1279005 edjee@google.com 2013-04-04 19:43:34 +00:00
  • 835dbf4516 Fix no received audio in tests. pwestin@webrtc.org 2013-04-04 17:24:15 +00:00
  • aa527bbc91 Disabling MixingTests due to race conditions. henrika@webrtc.org 2013-04-04 15:19:10 +00:00
  • fcb7c38b15 Two more sleep calls converted to use SleepMs(). This is CL 753005 in its new home. hta@webrtc.org 2013-04-04 08:56:34 +00:00
  • bb8ada686e TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC henrika@webrtc.org 2013-04-04 08:39:09 +00:00
  • 0c45957e3a Remove UDP transport API from VoE Review URL: https://webrtc-codereview.appspot.com/1236004 pwestin@webrtc.org 2013-04-03 15:43:57 +00:00
  • 0746ce1465 Fixes memory leak in AudioLevel class reported by memory try bots. henrika@webrtc.org 2013-04-03 11:58:12 +00:00
  • d108a46206 Fixes data race in WebRTCAudioDeviceTest.StartRecording reported by ThreadSanitizer henrika@webrtc.org 2013-04-03 11:25:31 +00:00
  • 82dcc9ff11 Remove UDP transport API from ViE Review URL: https://webrtc-codereview.appspot.com/1232004 pwestin@webrtc.org 2013-04-02 20:37:14 +00:00
  • 7b859cc1e9 Webrtc_Word32 => int32_t in video_coding/main/ pbos@webrtc.org 2013-04-02 15:54:38 +00:00
  • cfc07c943f Revert of r3747. henrike@webrtc.org 2013-04-02 14:55:44 +00:00
  • 95d88735ee Two more sleep calls converted to use SleepMs(). hta@webrtc.org 2013-04-02 14:46:33 +00:00
  • 4ff956f428 Fixes data race in WebRTCAudioDeviceTest.Construct reported by ThreadSanitizer henrika@webrtc.org 2013-04-02 11:59:11 +00:00
  • 46e626d3b8 Fix gflags compile error on x86 Android kjellander@webrtc.org 2013-04-02 11:07:04 +00:00
  • f81fad6267 Fix opus bitrate truncated to 16-bit int. This prevented setting bitrates higher than 2^16kbps. Review URL: https://webrtc-codereview.appspot.com/1275004 justinlin@chromium.org 2013-04-01 22:25:11 +00:00
  • 747c4cc96e For VGA (640x360), currently 1 thread is used. This change increases it to 2 threads. For HD, 4 threads are enabled. BUG=none TEST=run a hangout and screencast high framerate, high resolution windows of youtube. Observe that 1 cpu is insufficient to maintain high framerate with complex content. Review URL: https://webrtc-codereview.appspot.com/1203006 fbarchard@google.com 2013-04-01 22:16:45 +00:00
  • 65243bdb18 Updated Webrtc version to 3.28 Review URL: https://webrtc-codereview.appspot.com/1272006 elham@webrtc.org 2013-04-01 16:17:26 +00:00
  • 7f6b7cbcfc Revert r3743. marpan@webrtc.org 2013-03-29 21:35:22 +00:00
  • e882a47c8d Roll libvpx to 191157. -Pick up the libvpx roll to 8015a9ae. marpan@webrtc.org 2013-03-29 21:16:24 +00:00
  • 29f34b8727 Fix for issue: https://code.google.com/p/webrtc/issues/detail?id=1549 Review URL: https://webrtc-codereview.appspot.com/1270004 marpan@webrtc.org 2013-03-28 18:57:46 +00:00
  • 626c663115 Fixes build break in previous cl (https://code.google.com/p/webrtc/source/detail?r=3739) found by Android bots. henrike@webrtc.org 2013-03-28 16:31:51 +00:00
  • 93bea51517 Removed CPU APIs from VoEHardware. Code is now only used by test applications. henrike@webrtc.org 2013-03-28 15:58:49 +00:00
  • a442d4d983 Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. solenberg@webrtc.org 2013-03-28 09:14:36 +00:00
  • 80fccc29de Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..." wu@webrtc.org 2013-03-27 23:38:21 +00:00
  • 4c138e8fca Removed CPU APIs from VoEHardware. Code is now only used by test applications. henrike@webrtc.org 2013-03-27 21:23:42 +00:00
  • 458194ba65 Fix broken audio. leozwang@webrtc.org 2013-03-27 20:55:54 +00:00
  • 4b1cd5c5c0 G722-stereo has been missing when creating AudioDecoder. Review URL: https://webrtc-codereview.appspot.com/1266004 turaj@webrtc.org 2013-03-27 20:42:48 +00:00
  • 4d06db557a NetEq4 fails if the first packets inserted in are out-of-band DTMFs. turaj@webrtc.org 2013-03-27 18:31:42 +00:00
  • e1a7193869 Fix flakiness in network up/down event tests when running under memcheck. stefan@webrtc.org 2013-03-27 17:01:48 +00:00
  • add50b94a5 WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14. (required bumping minSdkVersion to 14) fischman@webrtc.org 2013-03-27 16:48:34 +00:00
  • bfacda60be Add interface to signal a network down event. stefan@webrtc.org 2013-03-27 16:36:01 +00:00
  • 686001dd96 Split condition_variable_win.cc into native (for Vista and newer OS versions) and generic implementation (based on events). henrike@webrtc.org 2013-03-26 14:16:05 +00:00
  • 1b31c78e5f Remove VoE's default call in Trace::SetLevelFilter. andrew@webrtc.org 2013-03-26 14:09:48 +00:00
  • d8a6e72057 Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust. solenberg@webrtc.org 2013-03-26 14:02:30 +00:00
  • 0633cccb4f Alphabetize include order in fake_voe_external_media.h. andrew@webrtc.org 2013-03-26 01:57:24 +00:00
  • 0e3077ab1f Restart Android capture after orientation change. Also prevent an NPE on exit. fischman@webrtc.org 2013-03-25 22:08:51 +00:00
  • c83a00ad49 Add some VoE and AudioProcessing mocks. andrew@webrtc.org 2013-03-25 21:20:38 +00:00
  • b87cc85beb Refactor unittest trace printouts to a separate class. andrew@webrtc.org 2013-03-25 16:23:37 +00:00
  • b4c441a785 Enable the below APIs for iOS. sjlee@webrtc.org 2013-03-25 11:12:20 +00:00
  • 7b48cedc57 libyuv r618 roll. Includes new psnr tool for evaluating codec quality. BUG=none TEST=none Review URL: https://webrtc-codereview.appspot.com/1241005 fbarchard@google.com 2013-03-24 02:58:00 +00:00
  • db4185664c Introduced pause and resume to the pacer Review URL: https://webrtc-codereview.appspot.com/1217007 pwestin@webrtc.org 2013-03-22 23:39:29 +00:00
  • 14c9909ef6 Updated WebRTC version to 3.27 Review URL: https://webrtc-codereview.appspot.com/1235004 elham@webrtc.org 2013-03-22 21:59:19 +00:00
  • a078d5cc38 Bugfix for extended RTP/RTCP test pwestin@webrtc.org 2013-03-22 20:03:03 +00:00
  • 26e35e1d06 Move the VIE tests to use external transport instead of the built in udp transport Review URL: https://webrtc-codereview.appspot.com/1216010 pwestin@webrtc.org 2013-03-22 19:21:27 +00:00
  • c1ffd337f1 Add trace printouts to all unit tests. andrew@webrtc.org 2013-03-22 17:13:23 +00:00
  • 94bc4cf905 Add min and target bitrate to VideoCodec. Review URL: https://webrtc-codereview.appspot.com/1214004 marpan@webrtc.org 2013-03-22 17:13:08 +00:00
  • e30823911c Move the VoE tests to use external transport instead of the built in udp transport Review URL: https://webrtc-codereview.appspot.com/1223006 pwestin@webrtc.org 2013-03-22 16:12:57 +00:00
  • e86f43b02a Roll Opus 1.0.2 tina.legrand@webrtc.org 2013-03-22 12:39:54 +00:00
  • 3ed599adb5 Bandwidth stats display in constraints-and-stats. hta@webrtc.org 2013-03-22 08:48:16 +00:00
  • 999e900fb6 Creating a copy of Udp transport under webrtc/test pwestin@webrtc.org 2013-03-21 16:38:05 +00:00
  • 2cec0b1670 Cleanup nanosleep -> SleepMs Remove some leftover stuff hta@webrtc.org 2013-03-21 14:02:29 +00:00
  • ae4e2b352b WebRtc_Word -> stdint in audio_coding/g711/ pbos@webrtc.org 2013-03-21 13:38:29 +00:00