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  • e07710cc91 Make SendCodec() lock-free. tommi@webrtc.org 2015-02-19 17:43:25 +00:00
  • be29b3b4c6 I420VideoFrame: Remove functions set_width, set_height, and ResetSize magjed@webrtc.org 2015-02-19 15:34:55 +00:00
  • be96bfb179 Re-land "Switch to using AudioEncoderIsac instead of ACMISAC" kwiberg@webrtc.org 2015-02-19 15:10:20 +00:00
  • 1ed6224eaf Revert r8430 "Remove dead stats from Video{Sender,Receiver}Info." pbos@webrtc.org 2015-02-19 13:57:03 +00:00
  • 287755246a Fix a problem with reading uninitialized memory in ACM henrik.lundin@webrtc.org 2015-02-19 13:55:36 +00:00
  • 8ad05b7628 Remove dead stats from Video{Sender,Receiver}Info. pbos@webrtc.org 2015-02-19 13:00:32 +00:00
  • 1d0fa5d352 Add RtcpPacketTypeCounter stats to new API. pbos@webrtc.org 2015-02-19 12:47:00 +00:00
  • 50604128db Method WebRtc_g722_encode that is eventually called always returns non-negative integer (internal counter) jmarusic@webrtc.org 2015-02-19 12:16:27 +00:00
  • 47d657b68e Remove Set/Get sending status from the default RTP module. mflodman@webrtc.org 2015-02-19 10:29:32 +00:00
  • 32c784c266 ViEExternalRendererImpl: Remove dependency to webrtc::VideoFrame magjed@webrtc.org 2015-02-19 10:03:52 +00:00
  • 3db042e2f0 Stop AndroidVideoCapturer asynchronously. The purpose is to avoid a deadlock between the C++ thread calling Stop and the Java thread that provides video frames. perkj@webrtc.org 2015-02-19 08:43:38 +00:00
  • 254840692e Add empty files to implement a in-memory DTLS identity store without breaking Chromium build. jiayl@webrtc.org 2015-02-18 23:58:16 +00:00
  • 652bc37a07 Adding two new stats to StatsReport. minyue@webrtc.org 2015-02-18 23:50:46 +00:00
  • a744a28b92 Templatize and clean up codec wildcards. jlmiller@webrtc.org 2015-02-18 21:37:46 +00:00
  • 30540fe722 Initialize RTPVideoHeader fields to correctly set simulcastIdx for non VP8 codecs. glaznev@webrtc.org 2015-02-18 20:30:03 +00:00
  • 9dfe7aac2e Fix WebRTC IP leaks. guoweis@webrtc.org 2015-02-18 20:27:17 +00:00
  • 931e0cf4b1 Fix WebRTC IP leaks. guoweis@webrtc.org 2015-02-18 19:09:42 +00:00
  • f358aea7bf Fix WebRTC IP leaks. guoweis@webrtc.org 2015-02-18 18:44:01 +00:00
  • 18c92472df Move Android MediaCodec encoder and decoder factories to separate files. glaznev@webrtc.org 2015-02-18 18:42:55 +00:00
  • 88828e77d9 Fix I420VideoFrame unittests magjed@webrtc.org 2015-02-18 15:53:39 +00:00
  • c0bd7be0df Adding two new stats to VoiceReceiverInfo minyue@webrtc.org 2015-02-18 15:24:13 +00:00
  • 8fbdcfd73f Revert "Switch default color format to YV12." perkj@webrtc.org 2015-02-18 15:19:39 +00:00
  • b255865e6e The PCM codecs can never fail, so we don't need to check the return value jmarusic@webrtc.org 2015-02-18 15:02:27 +00:00
  • 78619e2714 Revert of r8378 "Switch to using AudioEncoderIsac instead of ACMISAC" henrik.lundin@webrtc.org 2015-02-18 14:50:57 +00:00
  • 1c3e728aa9 Switch default color format to YV12. Currently N21 is used per default. But according to http://developer.android.com/reference/android/graphics/ImageFormat.html#YV12 YV12 has been mandatory to support since api level 12. Since YV12 and I420 is the same except for the order of planes, this format is cheaper to use. perkj@webrtc.org 2015-02-18 13:16:17 +00:00
  • 635838bd9b Re-implementing AcmOpusTest as AcmGenericCodecOpusTest henrik.lundin@webrtc.org 2015-02-18 13:15:08 +00:00
  • f68e186de3 Remove EnableMirroring and MirrorRenderStream magjed@webrtc.org 2015-02-18 12:54:48 +00:00
  • 131bea89d6 Offline screenshare quality test, plus loopback. sprang@webrtc.org 2015-02-18 12:46:06 +00:00
  • 0521127779 AudioEncoder: Rename virtual accessors to CamelCase kwiberg@webrtc.org 2015-02-18 12:00:32 +00:00
  • cc483b7379 Roll chromium_revision 601e6f3..b0c3ed3 (315263:316737) kjellander@webrtc.org 2015-02-18 10:37:51 +00:00
  • b4987bfc24 Send black frame with previous size when muting. pbos@webrtc.org 2015-02-18 10:13:09 +00:00
  • 7d721eea14 Adding speech_expand_rate to NetEQ Network Statistics. minyue@webrtc.org 2015-02-18 10:01:53 +00:00
  • 3864363e2c cricket::VideoFrame: Refactor CopyToBuffer into base class magjed@webrtc.org 2015-02-18 09:19:20 +00:00
  • dd4a8da68a Remove DISABLE_YUV flag magjed@webrtc.org 2015-02-18 08:47:13 +00:00
  • 97aaf68fed Bump to version 42. jansson@webrtc.org 2015-02-18 08:19:50 +00:00
  • bfa3c7253f Don't call g_thread_init on glib >=2.31.0 decurtis@webrtc.org 2015-02-17 21:22:48 +00:00
  • e9facf8bb3 Add range checks in a variety of places where the values will subsequently be expected to be 0-127. pkasting@chromium.org 2015-02-17 20:36:28 +00:00
  • 27669f320b Apply good settings to Beamformer aluebs@webrtc.org 2015-02-17 19:24:28 +00:00
  • b08f4045ec Fix issue 4061. guoweis@webrtc.org 2015-02-17 19:00:42 +00:00
  • 0abc6011b9 Remove SetCaptureDelay from the RTP module. mflodman@webrtc.org 2015-02-17 16:36:08 +00:00
  • 7663684258 Implement the Nada rmcat proposal within the simulation framework. stefan@webrtc.org 2015-02-17 16:03:45 +00:00
  • 71b35a4ce4 iLBC: Use uint8_t[] for byte arrays jmarusic@webrtc.org 2015-02-17 16:02:18 +00:00
  • 640313ce4f WebRtcVideoCapturer: Remove dead code |OnIncomingCapturedEncodedFrame| magjed@webrtc.org 2015-02-17 15:10:32 +00:00
  • 7a91acb94a ViECapturer: Remove unimplemented function declaration |DeliverCodedFrame| magjed@webrtc.org 2015-02-17 14:56:39 +00:00
  • 1a38a51119 Add default implementation to VideoSourceInterface of Stop and Restart. This is to make sure Chrome does not break when rolling. This should be reverted once Chrome has been updated. perkj@webrtc.org 2015-02-17 14:51:12 +00:00
  • a28a91d2f0 Fix data race for RTCPReceiver stats callback. pbos@webrtc.org 2015-02-17 14:45:08 +00:00
  • 8f605e8911 Add VideoSource::Stop and Restart methods. The purpose is to make sure that start and stop is called on the correct thread on Android. It also cleans up the Java VideoSource implementation. perkj@webrtc.org 2015-02-17 13:53:56 +00:00
  • 959dac7498 VideoCaptureImpl: Remove unused member variable |_capture_encoded_frame| magjed@webrtc.org 2015-02-17 13:44:25 +00:00
  • 4dd40d6b88 Signal threads for faster receiver destruction. pbos@webrtc.org 2015-02-17 13:22:43 +00:00
  • 0a7d4eed98 Remove usage of BitrateController in VoiceEngine. mflodman@webrtc.org 2015-02-17 12:57:14 +00:00
  • f9b5c1b3d0 Removing CELT. minyue@webrtc.org 2015-02-17 12:36:41 +00:00
  • 2c1bcf2cb4 Adding decoded_fec_rate to NetEQ Network Statistics. minyue@webrtc.org 2015-02-17 10:17:09 +00:00
  • 290cb56dca Remove TimeToSendPacket and TimeToSendPadding from the default module. mflodman@webrtc.org 2015-02-17 10:15:06 +00:00
  • c0fc4dd87c Add 'mac_x64' trybot to default set. kjellander@webrtc.org 2015-02-17 08:30:29 +00:00
  • 86196c4f48 Setup encoders inexpensively before first frame. pbos@webrtc.org 2015-02-16 21:02:00 +00:00
  • 34509d9f33 Fix an issue with comfort noise in ACMGenericCodecWrapper henrik.lundin@webrtc.org 2015-02-16 16:02:17 +00:00
  • e9f0f591b5 Enable bitrate probing by default in PacedSender. stefan@webrtc.org 2015-02-16 15:47:51 +00:00
  • fbc347f2ef Re-land r8342 "Switch to using AudioEncoderIsac instead of ACMISAC"" henrik.lundin@webrtc.org 2015-02-16 14:28:20 +00:00
  • ce22f13f0e GN: Changes for vp9, opus and direct trace kjellander@webrtc.org 2015-02-16 12:47:20 +00:00
  • e35fa96cbe Move isacfix.gypi and isac.gypi kjellander@webrtc.org 2015-02-16 12:46:41 +00:00
  • 0200f70792 Set webrtc_rtp category to be disabled by default. sprang@webrtc.org 2015-02-16 12:06:00 +00:00
  • 14b0279416 Break out code from bloated files in the BWE simulator. stefan@webrtc.org 2015-02-16 12:02:20 +00:00
  • 0f7f161ed6 Add audio_coding module OWNERS file. kjellander@webrtc.org 2015-02-16 09:53:09 +00:00
  • 4dc0003bed Revert r8342 "Switch to using AudioEncoderIsac instead of ACMISAC" henrik.lundin@webrtc.org 2015-02-14 09:42:26 +00:00
  • 30142bbe07 Add arraysize to overrides to avoid macros redefinitions in Chromium aluebs@webrtc.org 2015-02-14 02:45:57 +00:00
  • d3b453be17 Remove the incremental IP address behavior from virtualsocketserver guoweis@webrtc.org 2015-02-14 00:43:41 +00:00
  • 3341b401cc Fix bug parsing media descriptions: the final field isn't a codec type for any of DTLS/SCTP, SCTP, or SCTP/DTLS. pthatcher@webrtc.org 2015-02-13 21:14:22 +00:00
  • 92a19bcbd7 Simplify mask calculation aluebs@webrtc.org 2015-02-13 19:37:38 +00:00
  • 56cb0ea99c Add support for bi-directional simulations by having both an uplink and a downlink. stefan@webrtc.org 2015-02-13 15:46:23 +00:00
  • d5ce2e63df Remove EventWrapper::Reset(). pbos@webrtc.org 2015-02-13 14:58:18 +00:00
  • 5a7dc39277 This is a code clean up. No functional change intended. guoweis@webrtc.org 2015-02-13 14:31:26 +00:00
  • a8cc3440b1 Allowing RED decoding for Opus. minyue@webrtc.org 2015-02-13 14:01:54 +00:00
  • 96e4db9bea Split peerconnection_jni.cc into separate files. For now: java_helpers - JNI convenience functions etc. Can in theory be moved to libjingle / webrtc general one day. classreferenceholder - app/webrtc specific Java class loader. androidvideocapturer_jni - the jni part of the video capturer I added. peerconnection_jni - all the rest. perkj@webrtc.org 2015-02-13 12:46:51 +00:00
  • 8db5854eb0 Fix potential flakiness in voe_auto_test. solenberg@webrtc.org 2015-02-13 12:19:10 +00:00
  • 2adf4c4edd Re-enable BWE tests using baseline files. solenberg@webrtc.org 2015-02-13 12:04:53 +00:00
  • 58f6f01acc WebRTC now compiles for enable_android_opensl=1. henrika@webrtc.org 2015-02-13 11:35:42 +00:00
  • 40fdb8ab96 Remove flaky test cases from peerconnection_unittest. The chain of API calls is tested from top to bottom anyway. solenberg@webrtc.org 2015-02-13 11:09:20 +00:00
  • ba97ea69f0 audio_coding/codec/ilbc: Removed usage of macro WEBRTC_SPL_MUL_16_16 bjornv@webrtc.org 2015-02-13 09:51:40 +00:00
  • 2bd299a172 Remove call to RtpRtcp::RegisterSendPayload for the default RTP module. mflodman@webrtc.org 2015-02-13 09:52:01 +00:00
  • 40367f984b Remove default video encoders for new video API. pbos@webrtc.org 2015-02-13 08:00:06 +00:00
  • 94eb9a6005 Whitespace change to test gsubtreed. kjellander@webrtc.org 2015-02-13 07:40:20 +00:00
  • e388c19a9f Fix start bitrate settings for VP9 codec in AppRTCDemo. glaznev@webrtc.org 2015-02-13 00:34:30 +00:00
  • bb1219eca3 Add a unit test for callbacks with empty frames and fix bug in code henrik.lundin@webrtc.org 2015-02-12 15:53:25 +00:00
  • e01264306b Remove temporary GYP targets kjellander@webrtc.org 2015-02-12 13:38:10 +00:00
  • aafbec15f9 Remove ViENetwork::SetBandwidthEstimationConfig() interface since dynamically changing BWE settings isn't necessary now that AIMD is the default. solenberg@webrtc.org 2015-02-12 13:20:39 +00:00
  • 503c33666f Re-enabling LocalP2PTestAnswerVideo and LocalP2PTestAnswerAudio test cases in peerconnection_unittest. solenberg@webrtc.org 2015-02-12 13:12:50 +00:00
  • a9eaeebc6a Fix problem where Android VoE can not record on multiple channels. perkj@webrtc.org 2015-02-12 12:33:10 +00:00
  • 7c4d20fd6c Remove potential deadlock in RTPSenderAudio. pbos@webrtc.org 2015-02-12 12:20:08 +00:00
  • ff689be3c0 Use std::min and std::max instead of self-defined functions such as rtc::_min/_max. andresp@webrtc.org 2015-02-12 11:54:26 +00:00
  • 9e4e524f38 Use an external-only VideoRenderModule in Call. pbos@webrtc.org 2015-02-12 10:48:23 +00:00
  • a4ef2ce29d Remove getting max payload length from default module. mflodman@webrtc.org 2015-02-12 09:54:18 +00:00
  • 006521d5bd Makes libjingle_peerconnection_android_unittest run on networkless devices. phoglund@webrtc.org 2015-02-12 09:23:59 +00:00
  • 3ee4fe5a94 Re-land: Add API to get negotiated SSL ciphers pthatcher@webrtc.org 2015-02-11 22:34:36 +00:00
  • 76b4ac96cd Switch to using AudioEncoderIsac instead of ACMISAC henrik.lundin@webrtc.org 2015-02-11 21:37:07 +00:00
  • 6c68c85b46 Switch to using AudioEncoderOpus instead of ACMOpus henrik.lundin@webrtc.org 2015-02-11 21:33:34 +00:00
  • 1226e926e6 CVO capturer feature: allow unrotated frame flows through the capture pipeline. guoweis@webrtc.org 2015-02-11 18:37:54 +00:00
  • dc7b02277c CVO capturer feature: allow unrotated frame flows through the capture pipeline. guoweis@webrtc.org 2015-02-11 18:05:12 +00:00
  • 20e8f22766 CVO capturer feature: allow unrotated frame flows through the capture pipeline. guoweis@webrtc.org 2015-02-11 17:51:17 +00:00
  • 073dd7b423 WebRtc_GetCPUFeaturesARM is only available on android andrew@webrtc.org 2015-02-11 17:02:52 +00:00
  • a98e796615 Remove default RTP module functionality for setting CSRC. mflodman@webrtc.org 2015-02-11 15:45:56 +00:00