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  • f1a605cad6 Update DEPS to support Mac clang build. andrew@webrtc.org 2011-10-21 15:29:16 +00:00
  • 5eb64f06be Fix BitrateSent() API when having a default RTP module. stefan@webrtc.org 2011-10-21 13:42:50 +00:00
  • 158f496030 Fixes a rate control bug in the VP8 wrapper. stefan@webrtc.org 2011-10-21 13:15:16 +00:00
  • aa32319046 Implement unittest for proxies of MediaStreamTrackInterface and MediaStreamInterface. This cl also change MediaStreamProxy to only allow setting the state from the signaling thread. perkj@webrtc.org 2011-10-21 09:32:38 +00:00
  • ca8b3a387e kind() method in track interface is changed to std::string to keep uniformity with other get methods mallinath@webrtc.org 2011-10-21 09:18:25 +00:00
  • 96ba19034c ref_count.h file name changed to refcount.h to keep as other ( most ) files are named in libjingle. Review URL: http://webrtc-codereview.appspot.com/240008 mallinath@webrtc.org 2011-10-21 08:01:11 +00:00
  • ead87b5051 Fix potential issue where frame buffers might be freed while being decoded. stefan@webrtc.org 2011-10-21 06:46:37 +00:00
  • 2b0f094c8f Avoid reallocating the decodedImage for every decoded frame. stefan@webrtc.org 2011-10-21 06:39:03 +00:00
  • ee3dfa6f43 Review URL: http://webrtc-codereview.appspot.com/241007 mikhal@webrtc.org 2011-10-21 00:46:09 +00:00
  • 1af915d8ae video_coding: vp8: Updating error propagation threshold Review URL: http://webrtc-codereview.appspot.com/246002 mikhal@webrtc.org 2011-10-20 18:19:18 +00:00
  • 11330b003e Added myself to rtp module watch Review URL: http://webrtc-codereview.appspot.com/243003 pwestin@webrtc.org 2011-10-20 17:54:20 +00:00
  • d75889e2eb Change of Android makefiles to build latest video coding code. Review URL: http://webrtc-codereview.appspot.com/239008 kma@webrtc.org 2011-10-20 16:28:56 +00:00
  • 7cf893743a git-svn-id: http://webrtc.googlecode.com/svn/trunk@785 4adac7df-926f-26a2-2b94-8c16560cd09d henrika@webrtc.org 2011-10-20 12:30:35 +00:00
  • cedbb036d1 [Issue 101] Solves memory leak on Windows henrika@webrtc.org 2011-10-20 12:11:45 +00:00
  • 2ebc9ce5a3 Fix broken PeerConnection Dev build. Fix MediaStreamHandler::CommitLocalStreams refactoring error. perkj@webrtc.org 2011-10-20 11:52:31 +00:00
  • c4d1983b7b Changes in rtp_format_vp8_unittest to match the changes in CL 774. stefan@webrtc.org 2011-10-20 08:19:34 +00:00
  • f553ec70c7 Notifier and RefCount interface and implementation class name changed according to the naming convention. Review URL: http://webrtc-codereview.appspot.com/241003 mallinath@webrtc.org 2011-10-20 06:24:24 +00:00
  • ae499a2ac8 Set correct codec info before sending frame to VCM. mflodman@webrtc.org 2011-10-20 05:55:46 +00:00
  • 81f25f9ff8 Fixing build errors on Windows platform. Minor changes... kjellander@webrtc.org 2011-10-19 20:06:56 +00:00
  • f3f2f6abdb * Add include_internal_video_capture and include_internal_video_render to include/exclude the internal VCM and VRM. * Split the WEBRTC_VIDEO_EXTERNAL_CAPTURE_AND_RENDER into WEBRTC_INCLUDE_INTERNAL_VIDEO_CAPTURE and WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER. * Add DummyDeviceInfo for the case when WEBRTC_INCLUDE_INTERNAL_VIDEO_CAPTURE is not defined. Review URL: http://webrtc-codereview.appspot.com/224005 wu@webrtc.org 2011-10-19 18:42:17 +00:00
  • 509c9c5d09 operator + is evaluated before ?: henrike@webrtc.org 2011-10-19 18:31:01 +00:00
  • 4df8c9a2ed Review URL: http://webrtc-codereview.appspot.com/243001 henrike@webrtc.org 2011-10-19 18:30:25 +00:00
  • 7ecdf585cb Enable chromium_code:1 in the Chrome build. Review URL: http://webrtc-codereview.appspot.com/240001 andrew@webrtc.org 2011-10-19 17:53:56 +00:00
  • ffd28f95c5 Request key frames to battle error propagation. stefan@webrtc.org 2011-10-19 15:55:39 +00:00
  • d0752c370d video_coding: Update to hybrid mode: Set FEC values for zero below a threshold. Review URL: http://webrtc-codereview.appspot.com/245001 mikhal@webrtc.org 2011-10-19 15:48:30 +00:00
  • c693bac6e7 Only start ViEPerformanceMonitor when needed. mflodman@webrtc.org 2011-10-19 13:40:58 +00:00
  • b5475d0076 vie_auto_test will now obey the Mac .mm rules for files including objective-c code. phoglund@webrtc.org 2011-10-19 10:59:39 +00:00
  • 4c636764b7 Updated the AEC delay logging to output values in ms. PB output updated. Review URL: http://webrtc-codereview.appspot.com/223003 bjornv@webrtc.org 2011-10-19 08:47:40 +00:00
  • cc412c1735 Remove second instance of ViE PerformanceMonitor. mflodman@webrtc.org 2011-10-19 08:27:30 +00:00
  • ce8813da4e Using id instead of name when setting Mac/QTKit capture device. mflodman@webrtc.org 2011-10-19 06:45:16 +00:00
  • 4d5d5c1267 Reorganize the audio_processing source. andrew@webrtc.org 2011-10-19 01:40:33 +00:00
  • 5d3bdf71ab Fix clang warnings in ViE autotest. Review URL: http://webrtc-codereview.appspot.com/239004 andrew@webrtc.org 2011-10-19 01:09:41 +00:00
  • 8fd93d4d96 Move DeliverCapturedFrame from private to protected. Review URL: http://webrtc-codereview.appspot.com/246001 wu@webrtc.org 2011-10-19 00:16:36 +00:00
  • 1305a1d05e Fix rendering in new PeerConnection API. perkj@webrtc.org 2011-10-18 11:54:46 +00:00
  • 52eddf7378 Made Tina, Andrew and Jan as OWNERS to entire common_audio and removed the sub-OWNERS files. Let me know if that's fine. Review URL: http://webrtc-codereview.appspot.com/225006 bjornv@webrtc.org 2011-10-18 07:57:04 +00:00
  • 5b15cfc6dd Fix BWE unit test build issue stefan@webrtc.org 2011-10-18 07:22:33 +00:00
  • 61f07c3184 I have made a small fix so it will execute properly from the default working directory location (trunk), finding its resource files. kjellander@webrtc.org 2011-10-18 06:54:58 +00:00
  • 5dedd0ee38 Handling of white-space in DataLog::Combine henrik.lundin@webrtc.org 2011-10-18 05:45:08 +00:00
  • 929789b528 vie_auto_test - moved custom call specific functions to be static, added video protect method to custom call amyfong@webrtc.org 2011-10-17 21:57:08 +00:00
  • 76aea651ff When _audioConfigured, should not try to use the _video. Review URL: http://webrtc-codereview.appspot.com/224004 wu@webrtc.org 2011-10-17 21:40:32 +00:00
  • 0d55c8f96d Adding peerconnection_unittest. Review URL: http://webrtc-codereview.appspot.com/226004 henrike@webrtc.org 2011-10-17 21:12:45 +00:00
  • 5cb3064642 The change will separate the media tracks based on media type. MediaStreamInterface currently will have list for audio and video. This way we don't need to check for the track type before converting to respective mediatrack. Review URL: http://webrtc-codereview.appspot.com/230003 mallinath@webrtc.org 2011-10-17 13:19:08 +00:00
  • 63257d4bd2 Implement proxy for both audio and video tracks. The purpose of the proxy is that all calls to MediaStreamTracks should be done on the signaling thread. perkj@webrtc.org 2011-10-17 11:39:09 +00:00
  • 3765bd2cc2 Added AEC delay logging metrics to VoE. Echo metrics and delay logging metrics are enabled simultaneously through the SetEcMetricsStatus(). Updated standard and extended VoE tests. bjornv@webrtc.org 2011-10-17 08:49:23 +00:00
  • f10ea31211 Add IncomingFrameI420 to ViEExternalCapture interface to take captured video frame buffer as 3 planes. Review URL: http://webrtc-codereview.appspot.com/219004 wu@webrtc.org 2011-10-14 17:16:04 +00:00
  • 14aaaf116a Some re-organization of the fec-uep code: updated protection modes, comments, and some variable/function re-naming. Review URL: http://webrtc-codereview.appspot.com/231001 marpan@webrtc.org 2011-10-14 16:28:02 +00:00
  • 55c39f0940 Add mallinath@webrtc.org and wu@webrtc.org as the capture owner for US office. Review URL: http://webrtc-codereview.appspot.com/230001 wu@webrtc.org 2011-10-14 15:34:19 +00:00
  • 58691ebb97 Remove the DestroyDeviceInfo for mac video capture. (This is missed in r731.) Review URL: http://webrtc-codereview.appspot.com/229001 wu@webrtc.org 2011-10-14 15:13:16 +00:00
  • d0bdab0128 Adding API to get sent total bitrate, FEC bitrate and NACK bitrate. stefan@webrtc.org 2011-10-14 14:24:54 +00:00
  • e698eb7e27 Make the sanity check test a little more robust, and add a README file. Review URL: http://webrtc-codereview.appspot.com/220006 hta@webrtc.org 2011-10-14 13:56:26 +00:00
  • 26c041673f Added more tests, fixed a bug and refactored. phoglund@webrtc.org 2011-10-14 13:00:20 +00:00
  • 2111d3b0b0 Removed the vad_const files and added the constants to the files where they are used. Having them in a separate file did not add anything in readability or conceptual overview. Review URL: http://webrtc-codereview.appspot.com/230004 bjornv@webrtc.org 2011-10-14 12:58:34 +00:00
  • a59d80db45 Updated fixed point output data file after changes in nsx. Verified bitexactness before that CL and the CLs afterwards towards the new file. Review URL: http://webrtc-codereview.appspot.com/213003 bjornv@webrtc.org 2011-10-14 12:16:43 +00:00
  • c01c358f54 session/phone/channel.cc updates after new push of libjingle revision. Review URL: http://webrtc-codereview.appspot.com/225003 mallinath@webrtc.org 2011-10-14 09:45:24 +00:00
  • ebc0a00197 One of Justin comment was to have XXXXInterface and XXXX, rather than XXXX and XXXXImpl. So here are the changes, i don't like to call some the classes as interfaces like MediaStreamTrackListInterface, but they fit the criteria to be called as interface. Review URL: http://webrtc-codereview.appspot.com/226001 mallinath@webrtc.org 2011-10-14 07:04:02 +00:00
  • 03a86998cd Fixes for build errors introduced most likely earlier today. Review URL: http://webrtc-codereview.appspot.com/228003 henrike@webrtc.org 2011-10-13 23:36:38 +00:00
  • 0c378112ec Define NO_SOUND_SYSTEM for chromium build. Review URL: http://webrtc-codereview.appspot.com/226002 wu@webrtc.org 2011-10-13 22:35:01 +00:00
  • ebc405d9c6 Remove the fakeportallocator from the libjingle.gyp. Review URL: http://webrtc-codereview.appspot.com/228001 wu@webrtc.org 2011-10-13 18:36:04 +00:00
  • 4ee906d297 When WEBRTC_VIDEO_ENGINE_FILE_API is not defined, disable the code in vie_file_impl.cc and vie_file_image.cc so that we can remove the libjpeg dependency. Also disable the auto test for the vie file api. Review URL: http://webrtc-codereview.appspot.com/227001 wu@webrtc.org 2011-10-13 17:56:38 +00:00
  • 5a3e20f678 Removed unused variables (build error) for test_fec. Review URL: http://webrtc-codereview.appspot.com/223001 marpan@webrtc.org 2011-10-13 16:59:24 +00:00
  • 6c2d7107ae * Update to use the new libjingle release. * Stop using any local mods for the default build (non-dev). Review URL: http://webrtc-codereview.appspot.com/224001 wu@webrtc.org 2011-10-13 16:58:50 +00:00
  • 1da1ce0da5 First implementation of simulcast, adds VP8 simulcast to video engine. Changed API to RTP module Expanded Auto test with a test for simulcast Made the video codec tests compile Added the vp8_simulcast files to this cl Added missing auto test file Review URL: http://webrtc-codereview.appspot.com/188001 pwestin@webrtc.org 2011-10-13 15:19:55 +00:00
  • 103f33b734 Changes after comments received from Justin and Harald. Few comments are not implemented like moving track implementation to base<> and then have child classes based on the type of track. Review URL: http://webrtc-codereview.appspot.com/217004 mallinath@webrtc.org 2011-10-13 14:31:20 +00:00
  • 7951e819af Simple utility method for finding the project root dir (to be used by tests loading resource files) kjellander@webrtc.org 2011-10-13 12:24:41 +00:00
  • 6a34d584b8 Implement MediaStreamProxy. This implements a proxy for MediaStreams and MediaStreamTracklists. perkj@webrtc.org 2011-10-13 08:48:43 +00:00
  • 4c059d87b3 Add metric for number of packets discarded by JB due to not being decodable stefan@webrtc.org 2011-10-13 07:35:37 +00:00
  • 77d7d5455e Replace the DestroyDeviceInfo with a virtual destructor. Review URL: http://webrtc-codereview.appspot.com/212005 wu@webrtc.org 2011-10-12 16:57:53 +00:00
  • 38e400a967 Adding native client test page to test loopback. The test page is the same as the previouse test page but exchange offer messagesto answer messages. perkj@webrtc.org 2011-10-12 12:59:13 +00:00
  • e5542a0af5 Add file record and play functions to voe_cmd_test, fix Play local file (path was incorrect) Fixed: 24. Play local file (audio_long16.pcm) New: 34. Record a PCM file 35. Play a previously recorded PCM file locally 36. Play a previously recorded PCM file as microphone Review URL: http://webrtc-codereview.appspot.com/209001 amyfong@webrtc.org 2011-10-11 20:30:56 +00:00
  • 6330cf2a14 Fixed ViE AutoTest trace file names to be consistent Fixed some space issues in vie_autotest_custom_call.cc Fixed incorrect default codec W&H for I420 in vie_autotest_custom_call.cc Added functionality to modify a running custom call. The following options were added: 0. Finished modifying custom call 1. Change Video Codec 2. Change Video Size by Common Resolutions 3. Change Video Size by Width & Height 4. Change Video Device 5. Record Incoming Call 6. Record Outgoing Call 7. Play File on Video Channel(Assumes you recorded incoming & outgoing call) 8. Print Call information amyfong@webrtc.org 2011-10-11 18:17:22 +00:00
  • ea89922b56 Add VideoCaptureFactory so that we don't need to expose VideoCaptureImpl. wu@webrtc.org 2011-10-11 17:13:51 +00:00
  • 199f4defd3 Rename all .cc files which include Objective-C headers to .mm. andrew@webrtc.org 2011-10-11 15:43:35 +00:00
  • a0258defd4 Fixes test build errors (warnings treated as errors) in system_wrappers. Review URL: http://webrtc-codereview.appspot.com/212003 henrike@webrtc.org 2011-10-11 14:49:27 +00:00
  • 26c9ff983e Add dummy implementation of DataLog::Combine method henrik.lundin@webrtc.org 2011-10-11 14:43:41 +00:00
  • 791eec7424 Add API to get the number of packets discarded by the video jitter buffer due to being too late. stefan@webrtc.org 2011-10-11 07:53:43 +00:00
  • 06887aebae Fixes two bugs when decoding with packet losses. stefan@webrtc.org 2011-10-10 14:17:46 +00:00
  • ed081a99a9 Print info about the local and remote resolution in the Windows client. Review URL: http://webrtc-codereview.appspot.com/212001 tommi@webrtc.org 2011-10-10 12:58:21 +00:00
  • 73ba4160f6 Fix OnClose(socket, NO_ERROR) compile error on Linux. Merge Peerconnection_client_dev with Peerconnection_client. perkj@webrtc.org 2011-10-10 11:15:35 +00:00
  • 1843664f2a DataLog: Changing from common_types to typedefs henrik.lundin@webrtc.org 2011-10-10 09:56:52 +00:00
  • f7b36a47c0 Fix bug in the server where a wait request was incorrectly handled. Change the assert macro on Windows to make it easier to debug. Review URL: http://webrtc-codereview.appspot.com/212002 tommi@webrtc.org 2011-10-10 09:51:52 +00:00
  • c0b2250b20 Fix the Windows build. Review URL: http://webrtc-codereview.appspot.com/213004 tommi@webrtc.org 2011-10-10 08:43:33 +00:00
  • 5a695d6094 Fix bug in the client that caused signaling messages to be dropped. tommi@webrtc.org 2011-10-10 08:16:26 +00:00
  • d855bd4d6f C wrapper for DataLog class henrik.lundin@webrtc.org 2011-10-10 08:06:17 +00:00
  • 6364d128a1 Fix a couple of build warnings. Review URL: http://webrtc-codereview.appspot.com/214004 tommi@webrtc.org 2011-10-10 08:04:59 +00:00
  • e95458c30a Started rewriting video_engine tests to use GUnit. phoglund@webrtc.org 2011-10-10 07:23:51 +00:00
  • c8c4deb0bb Fix Windows build. %zu isn't supported in the crt implementation tommi@webrtc.org 2011-10-09 18:32:17 +00:00
  • 5a945ecc28 A little upgrade to the HTML test page: tommi@webrtc.org 2011-10-07 13:23:11 +00:00
  • 25e0b8e3a0 Python output flag and keyframe interval flags. kjellander@webrtc.org 2011-10-07 07:52:00 +00:00
  • a31b254084 Python output flag and keyframe interval flags. kjellander@webrtc.org 2011-10-07 06:50:22 +00:00
  • 80dd19be0a vplib tests: Removing old and unused file and directories. Note that the convert_test and scale_test directories are also removed. Review URL: http://webrtc-codereview.appspot.com/208001 mikhal@webrtc.org 2011-10-06 22:57:06 +00:00
  • f6ab63c08a Update PeerConnection_client to open a video capture device. perkj@webrtc.org 2011-10-06 20:36:23 +00:00
  • bf54ef9bb7 Removed code under a non-existing define. Review URL: http://webrtc-codereview.appspot.com/193006 henrike@webrtc.org 2011-10-06 18:14:25 +00:00
  • 1a2933c71a Fixes a Valgrind warning triggering when the number of pending messages hit the limit. Review URL: http://webrtc-codereview.appspot.com/200002 henrike@webrtc.org 2011-10-06 17:55:56 +00:00
  • 2915f6fc44 Use proper printf size_t specifier to fix Linux 32-bit build. andrew@webrtc.org 2011-10-06 16:37:03 +00:00
  • b2d4921f3b Remove trailing whitespace in AudioDevice. andrew@webrtc.org 2011-10-06 16:34:36 +00:00
  • d6132f54d2 Review URL: http://webrtc-codereview.appspot.com/193007 mikhal@webrtc.org 2011-10-06 16:23:38 +00:00
  • 3a6d4f4268 Fix setting VideoCaptureModule and VideoRenderer for local and remote streams. perkj@webrtc.org 2011-10-06 16:10:10 +00:00
  • 35a1756502 First version of video quality measurement program and test framework. See https://docs.google.com/a/google.com/document/d/1w6Nrxw6yTg_sDu18Ux8oZPEMo5F_R-zt62udrmmTeOc/edit?hl=en_US for background, details and additional instructions on usage. kjellander@webrtc.org 2011-10-06 06:44:54 +00:00
  • 3ce62fcfe4 Move merge_libs targets to their own gyp. andrew@webrtc.org 2011-10-06 01:03:18 +00:00
  • af57de006a Some code style changes in audio_processing/ns/main/source/ by Astyle, with a little manual modification. Review URL: http://webrtc-codereview.appspot.com/201002 kma@webrtc.org 2011-10-05 23:36:01 +00:00