Log - 131bea89d6f3742e649be84c91f8fd6c43b62d28 - platform/external/webrtc

131bea8 Offline screenshare quality test, plus loopback. by sprang@webrtc.org · 9 years ago
0521127 AudioEncoder: Rename virtual accessors to CamelCase by kwiberg@webrtc.org · 9 years ago
cc483b7 Roll chromium_revision 601e6f3..b0c3ed3 (315263:316737) by kjellander@webrtc.org · 9 years ago
b4987bf Send black frame with previous size when muting. by pbos@webrtc.org · 9 years ago
7d721ee Adding speech_expand_rate to NetEQ Network Statistics. by minyue@webrtc.org · 9 years ago
3864363 cricket::VideoFrame: Refactor CopyToBuffer into base class by magjed@webrtc.org · 9 years ago
dd4a8da Remove DISABLE_YUV flag by magjed@webrtc.org · 9 years ago
97aaf68 Bump to version 42. by jansson@webrtc.org · 9 years ago
bfa3c72 Don't call g_thread_init on glib >=2.31.0 by decurtis@webrtc.org · 9 years ago
e9facf8 Add range checks in a variety of places where the values will subsequently be by pkasting@chromium.org · 9 years ago
27669f3 Apply good settings to Beamformer by aluebs@webrtc.org · 9 years ago
b08f404 Fix issue 4061. by guoweis@webrtc.org · 9 years ago
0abc601 Remove SetCaptureDelay from the RTP module. by mflodman@webrtc.org · 9 years ago
7663684 Implement the Nada rmcat proposal within the simulation framework. by stefan@webrtc.org · 9 years ago
71b35a4 iLBC: Use uint8_t[] for byte arrays by jmarusic@webrtc.org · 9 years ago
640313c WebRtcVideoCapturer: Remove dead code |OnIncomingCapturedEncodedFrame| by magjed@webrtc.org · 9 years ago
7a91acb ViECapturer: Remove unimplemented function declaration |DeliverCodedFrame| by magjed@webrtc.org · 9 years ago
1a38a51 Add default implementation to VideoSourceInterface of Stop and Restart. by perkj@webrtc.org · 9 years ago
a28a91d Fix data race for RTCPReceiver stats callback. by pbos@webrtc.org · 9 years ago
8f605e8 Add VideoSource::Stop and Restart methods. by perkj@webrtc.org · 9 years ago
959dac7 VideoCaptureImpl: Remove unused member variable |_capture_encoded_frame| by magjed@webrtc.org · 9 years ago
4dd40d6 Signal threads for faster receiver destruction. by pbos@webrtc.org · 9 years ago
0a7d4ee Remove usage of BitrateController in VoiceEngine. by mflodman@webrtc.org · 9 years ago
f9b5c1b Removing CELT. by minyue@webrtc.org · 9 years ago
2c1bcf2 Adding decoded_fec_rate to NetEQ Network Statistics. by minyue@webrtc.org · 9 years ago
290cb56 Remove TimeToSendPacket and TimeToSendPadding from the default module. by mflodman@webrtc.org · 9 years ago
c0fc4dd Add 'mac_x64' trybot to default set. by kjellander@webrtc.org · 9 years ago
86196c4 Setup encoders inexpensively before first frame. by pbos@webrtc.org · 9 years ago
34509d9 Fix an issue with comfort noise in ACMGenericCodecWrapper by henrik.lundin@webrtc.org · 9 years ago
e9f0f59 Enable bitrate probing by default in PacedSender. by stefan@webrtc.org · 9 years ago
fbc347f Re-land r8342 "Switch to using AudioEncoderIsac instead of ACMISAC"" by henrik.lundin@webrtc.org · 9 years ago
ce22f13 GN: Changes for vp9, opus and direct trace by kjellander@webrtc.org · 9 years ago
e35fa96 Move isacfix.gypi and isac.gypi by kjellander@webrtc.org · 9 years ago
0200f70 Set webrtc_rtp category to be disabled by default. by sprang@webrtc.org · 9 years ago
14b0279 Break out code from bloated files in the BWE simulator. by stefan@webrtc.org · 9 years ago
0f7f161 Add audio_coding module OWNERS file. by kjellander@webrtc.org · 9 years ago
4dc0003 Revert r8342 "Switch to using AudioEncoderIsac instead of ACMISAC" by henrik.lundin@webrtc.org · 9 years ago
30142bb Add arraysize to overrides to avoid macros redefinitions in Chromium by aluebs@webrtc.org · 9 years ago
d3b453b Remove the incremental IP address behavior from virtualsocketserver by guoweis@webrtc.org · 9 years ago
3341b40 Fix bug parsing media descriptions: the final field isn't a codec type for any of DTLS/SCTP, SCTP, or SCTP/DTLS. by pthatcher@webrtc.org · 9 years ago
92a19bc Simplify mask calculation by aluebs@webrtc.org · 9 years ago
56cb0ea Add support for bi-directional simulations by having both an uplink and a downlink. by stefan@webrtc.org · 9 years ago
d5ce2e6 Remove EventWrapper::Reset(). by pbos@webrtc.org · 9 years ago
5a7dc39 This is a code clean up. No functional change intended. by guoweis@webrtc.org · 9 years ago
a8cc344 Allowing RED decoding for Opus. by minyue@webrtc.org · 9 years ago
96e4db9 Split peerconnection_jni.cc into separate files. by perkj@webrtc.org · 9 years ago
8db5854 Fix potential flakiness in voe_auto_test. by solenberg@webrtc.org · 9 years ago
2adf4c4 Re-enable BWE tests using baseline files. by solenberg@webrtc.org · 9 years ago
58f6f01 WebRTC now compiles for enable_android_opensl=1. by henrika@webrtc.org · 9 years ago
40fdb8a Remove flaky test cases from peerconnection_unittest. The chain of API calls is tested from top to bottom anyway. by solenberg@webrtc.org · 9 years ago
ba97ea6 audio_coding/codec/ilbc: Removed usage of macro WEBRTC_SPL_MUL_16_16 by bjornv@webrtc.org · 9 years ago
2bd299a Remove call to RtpRtcp::RegisterSendPayload for the default RTP module. by mflodman@webrtc.org · 9 years ago
40367f9 Remove default video encoders for new video API. by pbos@webrtc.org · 9 years ago
94eb9a6 Whitespace change to test gsubtreed. by kjellander@webrtc.org · 9 years ago
e388c19 Fix start bitrate settings for VP9 codec in AppRTCDemo. by glaznev@webrtc.org · 9 years ago
bb1219e Add a unit test for callbacks with empty frames and fix bug in code by henrik.lundin@webrtc.org · 9 years ago
e012643 Remove temporary GYP targets by kjellander@webrtc.org · 9 years ago
aafbec1 Remove ViENetwork::SetBandwidthEstimationConfig() interface since dynamically changing BWE settings isn't necessary now that AIMD is the default. by solenberg@webrtc.org · 9 years ago
503c336 Re-enabling LocalP2PTestAnswerVideo and LocalP2PTestAnswerAudio test cases in peerconnection_unittest. by solenberg@webrtc.org · 9 years ago
a9eaeeb Fix problem where Android VoE can not record on multiple channels. by perkj@webrtc.org · 9 years ago
7c4d20f Remove potential deadlock in RTPSenderAudio. by pbos@webrtc.org · 9 years ago
ff689be Use std::min and std::max instead of self-defined functions such as rtc::_min/_max. by andresp@webrtc.org · 9 years ago
9e4e524 Use an external-only VideoRenderModule in Call. by pbos@webrtc.org · 9 years ago
a4ef2ce Remove getting max payload length from default module. by mflodman@webrtc.org · 9 years ago
006521d Makes libjingle_peerconnection_android_unittest run on networkless devices. by phoglund@webrtc.org · 9 years ago
3ee4fe5 Re-land: Add API to get negotiated SSL ciphers by pthatcher@webrtc.org · 9 years ago
76b4ac9 Switch to using AudioEncoderIsac instead of ACMISAC by henrik.lundin@webrtc.org · 9 years ago
6c68c85 Switch to using AudioEncoderOpus instead of ACMOpus by henrik.lundin@webrtc.org · 9 years ago
1226e92 CVO capturer feature: allow unrotated frame flows through the capture pipeline. by guoweis@webrtc.org · 9 years ago
dc7b022 CVO capturer feature: allow unrotated frame flows through the capture pipeline. by guoweis@webrtc.org · 9 years ago
20e8f22 CVO capturer feature: allow unrotated frame flows through the capture pipeline. by guoweis@webrtc.org · 9 years ago
073dd7b WebRtc_GetCPUFeaturesARM is only available on android by andrew@webrtc.org · 9 years ago
a98e796 Remove default RTP module functionality for setting CSRC. by mflodman@webrtc.org · 9 years ago
a6e8ceb Fix false positive DHECK in event_posix.cc by sprang@webrtc.org · 9 years ago
11426dc Don't rely on webrtc/base/scoped_ptr.h to include stuff for you by kwiberg@webrtc.org · 9 years ago
fbcb5ce Remove VideoSendStreamTest.ProducesStats. by pbos@webrtc.org · 9 years ago
9d94a0c Switch to QueueUserAPC for shutting down the thread (no event needed). by tommi@webrtc.org · 9 years ago
fddeaf5 Switch to using AudioEncoderG722 instead of ACMG722 by henrik.lundin@webrtc.org · 9 years ago
83bc721 Add Android specific VideoCapturer. by perkj@webrtc.org · 9 years ago
c18957e Make Git ignore in resources more fine-grained by kjellander@webrtc.org · 9 years ago
354becf Remove Git ignore exclusion of .sha1 files by kjellander@webrtc.org · 9 years ago
7cc92aa Use WebRtcVideoRenderFrame for texture frames. by pbos@webrtc.org · 9 years ago
62f6e75 Refactoring WebRTC Java/JNI audio recording in C++ and Java. by henrika@webrtc.org · 9 years ago
c2d0473 Switch to using AudioEncoderPcm16B instead of ACMPCM16B by henrik.lundin@webrtc.org · 9 years ago
f58fe0a Rename GYP and GN targets for video capture+render. by kjellander@webrtc.org · 9 years ago
2c29c2e C++ readability review for ajm. by andrew@webrtc.org · 9 years ago
5d60895 Fix bug when there are no blocks in a chunk in Beamformer by aluebs@webrtc.org · 9 years ago
bc35703 Add a method to remove an existing renderer from the internal list of Android renderers. by glaznev@webrtc.org · 9 years ago
bc40324 Merge fixes and changed for Android AppRTCDemo from internal repo. by glaznev@webrtc.org · 9 years ago
d35a5c3 Make ChannelBuffer aware of frequency bands by aluebs@webrtc.org · 9 years ago
d7472b5 base/arraysize.h: We use size_t, so need to include stddef.h by kwiberg@webrtc.org · 9 years ago
91ba79a Make sure that the norms are positive in Beamformer by aluebs@webrtc.org · 9 years ago
b6856d2 Apply mask smoothing in Beamformer by aluebs@webrtc.org · 9 years ago
8da96ac Switch to using AudioEncoderIlbc instead of ACMILBC by henrik.lundin@webrtc.org · 9 years ago
1a072f9 Address comments from previous review round for rtc::Event. by tommi@webrtc.org · 9 years ago
f4c10d2 Always use DeliverI420Frame in WebRtcVideoEngine. by pbos@webrtc.org · 9 years ago
027e113 Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig. by stefan@webrtc.org · 9 years ago
30015e3 Fix bug in EventPosix where we'd miss a set event. by tommi@webrtc.org · 9 years ago
648f5d6 pcm16b: Make input arrays const and use uint8_t[] for byte arrays by kwiberg@webrtc.org · 9 years ago
948d617 Create a separate thread for pacing. by mflodman@webrtc.org · 9 years ago