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  • 200ac007ef Remove temp files in audio_processing_unittest.cc. pbos@webrtc.org 2015-02-03 14:14:01 +00:00
  • 0e8bf6c4d3 Enable bitrate probing by default. stefan@webrtc.org 2015-02-03 12:33:51 +00:00
  • b1786dbab0 audio_processing: Added a new AEC delay metric value that gives the amount of poor delays bjornv@webrtc.org 2015-02-03 06:06:26 +00:00
  • 0e81fdf5d2 Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting. pkasting@chromium.org 2015-02-02 23:54:03 +00:00
  • 19f3f71c98 Fix apparent typo: int -> char. pkasting@chromium.org 2015-02-02 19:44:25 +00:00
  • 946ad76f7e Switched lists of packets to lists of packet pointers. Allows Packet polymorphism. stefan@webrtc.org 2015-02-02 14:51:20 +00:00
  • c957ffc6dc Fixed potential crash if rtp packet history is completely full. sprang@webrtc.org 2015-02-02 13:08:02 +00:00
  • c420a86f4c Change name for local CriticalSectionScoped variable henrik.lundin@webrtc.org 2015-02-02 10:36:30 +00:00
  • a1dfbf1e5c WebRtcG722_Decode: Input array should be const uint8_t[] kwiberg@webrtc.org 2015-02-02 08:58:03 +00:00
  • 026b892e72 Using << on an int8_t or uint8_t will output a character rather than a number. Places that do this need to cast to int to get the desired behavior. pkasting@chromium.org 2015-01-30 19:53:42 +00:00
  • 005b6fffe6 Convert some EXPECTs to ASSERTs to avoid crashes when object creation fails. pkasting@chromium.org 2015-01-30 19:41:42 +00:00
  • 5e161616b1 Remove CPU monitor from WebRtcVideoEngine2. pbos@webrtc.org 2015-01-30 15:31:03 +00:00
  • aef0779dab Rewrite ThreadWindows. tommi@webrtc.org 2015-01-30 15:06:10 +00:00
  • f2ec814e0f Move use of DEPTH into build_with_chromium==1. kjellander@webrtc.org 2015-01-30 14:54:36 +00:00
  • f88bee6d88 Refactor senders into senders and sources in the simulation framework. stefan@webrtc.org 2015-01-30 14:36:37 +00:00
  • a671f4b2cb Fixing a VoE test to set correct rate for iSAC henrik.lundin@webrtc.org 2015-01-30 13:04:29 +00:00
  • 05db352f56 Fix a bug in ACM test channel henrik.lundin@webrtc.org 2015-01-30 13:03:45 +00:00
  • 3154a1cf9d Reland r8210 "Add a new parameter to ACMGenericCodec constructor"" henrik.lundin@webrtc.org 2015-01-30 12:29:25 +00:00
  • 4455f6243a WebRtcIsacfix_Time2SpecNeon and _Spec2TimeNeon: Fix stack alignment henrik.lundin@webrtc.org 2015-01-30 11:58:00 +00:00
  • 8820ac7cc4 peerconnectin_server: missing comma in sprintfn() in r8128 braveyao@webrtc.org 2015-01-30 09:58:04 +00:00
  • 2bbc35d896 Remove unused method, SetAffinity, from the ThreadWrapper class. The method was also not consistently implemented across all platforms. tommi@webrtc.org 2015-01-30 09:42:37 +00:00
  • 6752b85ff7 Revert r8210 "Add a new parameter to ACMGenericCodec constructor" henrik.lundin@webrtc.org 2015-01-30 06:35:44 +00:00
  • c3643f2fe3 Add a new parameter to ACMGenericCodec constructor henrik.lundin@webrtc.org 2015-01-30 06:14:28 +00:00
  • 2444d9605a Control the max IPv6 Networks used by WebRTC. guoweis@webrtc.org 2015-01-30 00:09:28 +00:00
  • 4ddde2e3ad Add arbitrary microphone geometry input to audioproc_f test utility. mgraczyk@chromium.org 2015-01-29 22:39:44 +00:00
  • 13980253f0 Add new members to AudioEncoderOpus::Config henrik.lundin@webrtc.org 2015-01-29 16:08:40 +00:00
  • 7a37bfc240 Revert 8203 "Reducing locking in OveruseFrameDetector and increa..." tommi@webrtc.org 2015-01-29 16:08:20 +00:00
  • a33f05e8d7 Re-land "Remove <(webrtc_root) from source file entries." kjellander@webrtc.org 2015-01-29 14:29:45 +00:00
  • bdebccf384 Fix a number of things in AudioEncoderDecoderIsac* henrik.lundin@webrtc.org 2015-01-29 14:10:32 +00:00
  • 18e758526d Reducing locking in OveruseFrameDetector and increasing constness. tommi@webrtc.org 2015-01-29 12:34:40 +00:00
  • 50fe359eb6 Add tracing for slow paths in new video API. pbos@webrtc.org 2015-01-29 12:33:07 +00:00
  • 4161715e3f Remove ChangeUniqueID. tommi@webrtc.org 2015-01-29 12:12:49 +00:00
  • 1ece0cbbec Revert "Remove <(webrtc_root) from source file entries." kjellander@webrtc.org 2015-01-29 12:02:01 +00:00
  • a26f511dd2 Remove frame copy in ViEExternalRendererImpl::RenderFrame magjed@webrtc.org 2015-01-29 11:45:07 +00:00
  • a87c398a41 Move audio_codec_speed_tests into include_tests==1 condition. kjellander@webrtc.org 2015-01-29 10:39:07 +00:00
  • 2d2a1f9f05 Remove <(webrtc_root) from source file entries. kjellander@webrtc.org 2015-01-29 10:23:45 +00:00
  • 73ca1945ec Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h kwiberg@webrtc.org 2015-01-29 09:12:47 +00:00
  • 43c883954f Allow rtp packet history to dynamically expand in size. sprang@webrtc.org 2015-01-29 09:09:17 +00:00
  • 827d7e806a Change AsyncInvoker to store its closure in a scoped_refptr instead of using a raw pointer. perkj@webrtc.org 2015-01-29 08:53:45 +00:00
  • a742cb1f37 Enable DTLS for peerconnection example. If it's a loopback test, then we recreate another peerconnection with DTLS off. braveyao@webrtc.org 2015-01-29 04:23:01 +00:00
  • f17ee9c709 Add case to ApmTest.Process to test the extended filter mode aluebs@webrtc.org 2015-01-29 00:03:53 +00:00
  • e7a4a12f83 Add arraysize() macro from Chromium, and make use of it in a few places. pkasting@chromium.org 2015-01-28 21:36:55 +00:00
  • 035e9123e9 Move channel_buffer.{h,cc} to common_audio. kjellander@webrtc.org 2015-01-28 19:57:00 +00:00
  • a67ca1a3bb Only report the first rtp packet because it indicates the media has started flowing. BUG= R=juberti@webrtc.org honghaiz@google.com 2015-01-28 19:48:33 +00:00
  • a094cac11f Add stats for network merge. guoweis@webrtc.org 2015-01-28 19:34:05 +00:00
  • 7d2b6a9346 Enable Clang warning implicit-fallthrough and annotate the code. kjellander@webrtc.org 2015-01-28 18:37:58 +00:00
  • a907e01c63 Adding constness. tommi@webrtc.org 2015-01-28 17:33:12 +00:00
  • 664ccb7d8d Reland r8125: Modify some tests to never use DTX disable mode henrik.lundin@webrtc.org 2015-01-28 14:49:05 +00:00
  • 37c0559c1e Notify jitter buffer about received FEC packets (to avoid sending NACK request for these packets). asapersson@webrtc.org 2015-01-28 13:58:27 +00:00
  • 22c2f0572b Add "score" unit to SSIM perf score output. kjellander@webrtc.org 2015-01-28 13:52:08 +00:00
  • 4aecd008dd Add support for 40 and 60 ms frames to AudioEncoderIlbc henrik.lundin@webrtc.org 2015-01-28 13:16:31 +00:00
  • 2a6558c2a5 Make sure ByteReader<T>::Read* is properly constified. sprang@webrtc.org 2015-01-28 12:37:36 +00:00
  • 7aef80c6d1 GN: Remove webrtc_base target in favor for rtc_base. kjellander@webrtc.org 2015-01-28 07:55:26 +00:00
  • 9b64a6edd7 Adjust parameter in videoprocessor_integrationtest for VP9. marpan@webrtc.org 2015-01-27 23:59:01 +00:00
  • dc8a9da386 Adjust qp-max settinhg in VP9 wrapper. marpan@webrtc.org 2015-01-27 23:08:24 +00:00
  • 922cfcd150 Use non-zero data in AudioRingBufferTest. andrew@webrtc.org 2015-01-27 21:59:33 +00:00
  • 36401aba62 Update GAE API paths for join/leave. tkchin@webrtc.org 2015-01-27 21:34:39 +00:00
  • 8bb32d600b Minor updates to AudioEncoderCng henrik.lundin@webrtc.org 2015-01-27 20:53:56 +00:00
  • db1ebf6c0c Add jakehilton@gmail.com to AUTHORS tnakamura@webrtc.org 2015-01-27 19:15:11 +00:00
  • 478cedc055 Add new methods to AudioEncoder interface henrik.lundin@webrtc.org 2015-01-27 18:24:45 +00:00
  • 5614cf16e7 audio_processing: Use fixed aggregation window in delay metrics bjornv@webrtc.org 2015-01-27 18:09:52 +00:00
  • 6e251822cd Whitespace change after enabling gnumbd kjellander@webrtc.org 2015-01-27 16:46:02 +00:00
  • ccd608eeab Whitespace change for git updater kjellander@webrtc.org 2015-01-27 14:24:40 +00:00
  • 0bc73a1b72 Whitespace change to trigger git updater kjellander@webrtc.org 2015-01-27 14:13:13 +00:00
  • f68ffca050 Add PRESUBMIT check for GYP files including source files above itself. kjellander@webrtc.org 2015-01-27 13:13:24 +00:00
  • 76e5e207ad Roll chromium_revision 4664fe0..9070a80 (312733:313233) kjellander@webrtc.org 2015-01-27 13:11:10 +00:00
  • 273fbbb921 Update StreamDataCounter with FEC bytes. asapersson@webrtc.org 2015-01-27 12:17:29 +00:00
  • 70117a83d4 AEC: Implements a new function for calculating delay metrics bjornv@webrtc.org 2015-01-27 11:30:54 +00:00
  • fc5ad95fec Reland of: "Implement elapsed time and capture start NTP time estimation." revision @8139 magjed@webrtc.org 2015-01-27 09:57:01 +00:00
  • 8501ee632b Support VP8 HW decoding on devices with Exynos codec. glaznev@webrtc.org 2015-01-26 23:07:19 +00:00
  • df9a41d270 Fix bug in GetREDStatus(): it doesn't actually return the current status. pkasting@chromium.org 2015-01-26 22:35:29 +00:00
  • 82415e395f Update AppRTCDemo to use renamed GAE messages. glaznev@webrtc.org 2015-01-26 22:22:50 +00:00
  • 041035b390 Add an AudioRingBuffer class wrapper for the ring_buffer.h C interface. andrew@webrtc.org 2015-01-26 21:23:53 +00:00
  • 4dba2e98a2 Consolidate anonymous namespace content and file-static methods to all be in the anonymous namespace, in preparation for refactoring a few of the functions a little. pkasting@chromium.org 2015-01-26 19:59:32 +00:00
  • d7e34e1086 Make it easier to use external libyuv + cleanup GYP files. kjellander@webrtc.org 2015-01-26 19:17:26 +00:00
  • d25c034051 Refactor common_audio/vad: Removed usage of macro WEBRTC_SPL_MUL_16_16() bjornv@webrtc.org 2015-01-26 15:32:47 +00:00
  • 04cd466bd5 Move ThreadChecker into rtc_base_approved. tommi@webrtc.org 2015-01-26 15:27:29 +00:00
  • 38d11b8529 Enable encoder multi-threading for VP9. marpan@webrtc.org 2015-01-26 15:21:36 +00:00
  • 6f200b5b87 Temporarily revert r8147 ("Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h") kwiberg@webrtc.org 2015-01-26 13:03:32 +00:00
  • b6fab2b1cd Introduce rtc::CheckedDivExact henrik.lundin@webrtc.org 2015-01-26 11:08:53 +00:00
  • 19eb4e4b86 Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h kwiberg@webrtc.org 2015-01-26 08:57:57 +00:00
  • 995b4c9e8a Remove win_asan trybot from PRESUBMIT.py kjellander@webrtc.org 2015-01-25 19:27:03 +00:00
  • acb8085678 Roll chromium_revision c086b4e..4664fe0 (312108:312733) kjellander@webrtc.org 2015-01-25 19:17:56 +00:00
  • 7519de519e Revert 8136 "Remove frame copy in ViEExternalRendererImpl::Rende..." tkchin@webrtc.org 2015-01-23 21:20:41 +00:00
  • 0f98844749 Revert 8139 "Implement elapsed time and capture start NTP time e..." tkchin@webrtc.org 2015-01-23 21:17:38 +00:00
  • dacdd9403d Reland r7980: Accept incoming pings before remote answer is set, to reduce connection latency. Set ICE connection state to 'checking' after setting the remote answer, so that it can transition into 'connected' if the peer reflexive connection is up before any remote candidate is set. See more details in crbug/446908 jiayl@webrtc.org 2015-01-23 17:33:34 +00:00
  • 8919cfe9ce Change a GYP reference to cpufeatures.gypi fdegans@chromium.org 2015-01-23 16:35:17 +00:00
  • ad3ee2c46b Implement elapsed time and capture start NTP time estimation. pbos@webrtc.org 2015-01-23 14:55:00 +00:00
  • a02d76845f Disable DtmfSenderTest.InsertDtmfWithCommaAsDelay due to flakiness kjellander@webrtc.org 2015-01-23 14:34:52 +00:00
  • 456f01441a Re-allowing RED in voice engine. minyue@webrtc.org 2015-01-23 11:58:42 +00:00
  • 182ea46fac Remove frame copy in ViEExternalRendererImpl::RenderFrame magjed@webrtc.org 2015-01-23 11:50:13 +00:00
  • 73ee4537be Switch to use range based loops in the BWE simulation framework. stefan@webrtc.org 2015-01-23 08:29:52 +00:00
  • 36d5c3cb44 Leave BIO_METHOD non-const. davidben@webrtc.org 2015-01-22 23:06:17 +00:00
  • 586f2eda0d Change GetStreamBySsrc to not copy StreamParams. This is something I stumbled upon while looking at string copying we do (in spades) and did a simple change to not be constantly copying things around needlessly. There's a lot more that can be done in these files of course so this is sort of a reminder for future code edits that it's possible to design interfaces/function in a way that's more performance aware and avoid forcing creation of copies, while still being very simple. Also, we can use lambdas now :) tommi@webrtc.org 2015-01-22 23:00:41 +00:00
  • 7e5b380437 Fix a crash in AllocationSequence. Internal bug 19074679. jiayl@webrtc.org 2015-01-22 21:28:39 +00:00
  • ff108fe508 Revert 8125 "Modify some tests to never use DTX disable mode" kjellander@webrtc.org 2015-01-22 19:02:03 +00:00
  • b40c7bb53c Change sprintf use in talk samples to snprintf jlmiller@webrtc.org 2015-01-22 18:49:06 +00:00
  • ea1c84285c Correct GetDriveType error handling. jlmiller@webrtc.org 2015-01-22 17:44:19 +00:00
  • 043db24767 Modify some tests to never use DTX disable mode henrik.lundin@webrtc.org 2015-01-22 13:30:58 +00:00
  • e5251ad63c Integrate send-side BWE into simulation framework. stefan@webrtc.org 2015-01-22 10:10:53 +00:00