- 7227391 Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530) by kjellander@webrtc.org · 10 years ago
- b28474c Add H.264 HW encoder and decoder support for Android. by glaznev@webrtc.org · 10 years ago
- 77e11bb Wire up preferred/nominal_bitrate to stats. by pbos@webrtc.org · 10 years ago
- 829a6f4 Merge ACMGenericCodec and ACMGenericCodecWrapper by henrik.lundin@webrtc.org · 10 years ago
- f3a306b g722: Enhanced documentation. Added CHECK. by jmarusic@webrtc.org · 10 years ago
- 2acec4c Enhanced documentation. Replaced DCHECK with CHECK. by jmarusic@webrtc.org · 10 years ago
- 962c624 Refactoring WebRTC Java/JNI audio track in C++ and Java. by henrika@webrtc.org · 10 years ago
- 2ad3bb1 Reland patch for Switch default color format to YV12 on Android. by perkj@webrtc.org · 10 years ago
- 8278c07 Enable NACK under SendsAndReceivesH264. by pbos@webrtc.org · 10 years ago
- fa58745 Delete all codec-specific subclasses of ACMGenericCodec by henrik.lundin@webrtc.org · 10 years ago
- 2a5cfc2 Replaced unnecessary check with an explicit CHECK. by jmarusic@webrtc.org · 10 years ago
- 343096a Fix incorrect rtx config in full_stack tests. by sprang@webrtc.org · 10 years ago
- 1467421 Fix for flaky test: VideoSendStreamTest.RtcpSenderReportContainsMediaBytesSent. by asapersson@webrtc.org · 10 years ago
- 50e2816 Move SetTargetSendBitrates logic from default module to payload router. by mflodman@webrtc.org · 10 years ago
- a43fce6 Add functions rtc::AtomicOps::Load and rtc::RefCountedObject::HasOneRef by magjed@webrtc.org · 10 years ago
- 2af3057 Revert "When clearing the priority message queue, don't copy an item to itself." by decurtis@webrtc.org · 10 years ago
- 2bffc3c When clearing the priority message queue, don't copy an item to itself. by decurtis@webrtc.org · 10 years ago
- d3a487c Exclude end-to-end test RestartingSendStreamPreservesRtpStatesWithRt on memcheck. by marpan@webrtc.org · 10 years ago
- 3c4668e Amend CpuMonitor fix. by torbjorng@webrtc.org · 10 years ago
- f906e55 Add CpuMonitor to Android ApprtcDemo by torbjorng@webrtc.org · 10 years ago
- 7ac374a Fix shutdown race for ViEEncoder when there is a frame in the encoder. by mflodman@webrtc.org · 10 years ago
- dc77d74 Disable FullStackTest.ForemanCifPlr5 temporarily while investigating flakiness. by sprang@webrtc.org · 10 years ago
- ec45e3b Fix test race in GetStatsMultipleSendStreams. by pbos@webrtc.org · 10 years ago
- 804eb46 Change default from GICE to ICE5245 for SDP offers by jlmiller@webrtc.org · 10 years ago
- d3d3baa Copy SetThreadName from webrtc/base/thread.cc into thread_win.cc by tommi@webrtc.org · 10 years ago
- 661af50 Small Beamformer optimization by aluebs@webrtc.org · 10 years ago
- cce874b Fix libjingle_media_unittest codec comparison issue by guoweis@webrtc.org · 10 years ago
- bc6961f Make webrtc 50 KB smaller by not inlining Codec. by guoweis@webrtc.org · 10 years ago
- e07710c Make SendCodec() lock-free. by tommi@webrtc.org · 10 years ago
- be29b3b I420VideoFrame: Remove functions set_width, set_height, and ResetSize by magjed@webrtc.org · 10 years ago
- be96bfb Re-land "Switch to using AudioEncoderIsac instead of ACMISAC" by kwiberg@webrtc.org · 10 years ago
- 1ed6224 Revert r8430 "Remove dead stats from Video{Sender,Receiver}Info." by pbos@webrtc.org · 10 years ago
- 2877552 Fix a problem with reading uninitialized memory in ACM by henrik.lundin@webrtc.org · 10 years ago
- 8ad05b7 Remove dead stats from Video{Sender,Receiver}Info. by pbos@webrtc.org · 10 years ago
- 1d0fa5d Add RtcpPacketTypeCounter stats to new API. by pbos@webrtc.org · 10 years ago
- 5060412 Method WebRtc_g722_encode that is eventually called always returns non-negative integer (internal counter) by jmarusic@webrtc.org · 10 years ago
- 47d657b Remove Set/Get sending status from the default RTP module. by mflodman@webrtc.org · 10 years ago
- 32c784c ViEExternalRendererImpl: Remove dependency to webrtc::VideoFrame by magjed@webrtc.org · 10 years ago
- 3db042e Stop AndroidVideoCapturer asynchronously. by perkj@webrtc.org · 10 years ago
- 2548406 Add empty files to implement a in-memory DTLS identity store without breaking Chromium build. by jiayl@webrtc.org · 10 years ago
- 652bc37 Adding two new stats to StatsReport. by minyue@webrtc.org · 10 years ago
- a744a28 Templatize and clean up codec wildcards. by jlmiller@webrtc.org · 10 years ago
- 30540fe Initialize RTPVideoHeader fields to correctly set simulcastIdx for non VP8 codecs. by glaznev@webrtc.org · 10 years ago
- 9dfe7aa Fix WebRTC IP leaks. by guoweis@webrtc.org · 10 years ago
- 931e0cf Fix WebRTC IP leaks. by guoweis@webrtc.org · 10 years ago
- f358aea Fix WebRTC IP leaks. by guoweis@webrtc.org · 10 years ago
- 18c9247 Move Android MediaCodec encoder and decoder factories to separate files. by glaznev@webrtc.org · 10 years ago
- 88828e7 Fix I420VideoFrame unittests by magjed@webrtc.org · 10 years ago
- c0bd7be Adding two new stats to VoiceReceiverInfo by minyue@webrtc.org · 10 years ago
- 8fbdcfd Revert "Switch default color format to YV12." by perkj@webrtc.org · 10 years ago
- b255865 The PCM codecs can never fail, so we don't need to check the return value by jmarusic@webrtc.org · 10 years ago
- 78619e2 Revert of r8378 "Switch to using AudioEncoderIsac instead of ACMISAC" by henrik.lundin@webrtc.org · 10 years ago
- 1c3e728 Switch default color format to YV12. by perkj@webrtc.org · 10 years ago
- 635838b Re-implementing AcmOpusTest as AcmGenericCodecOpusTest by henrik.lundin@webrtc.org · 10 years ago
- f68e186 Remove EnableMirroring and MirrorRenderStream by magjed@webrtc.org · 10 years ago
- 131bea8 Offline screenshare quality test, plus loopback. by sprang@webrtc.org · 10 years ago
- 0521127 AudioEncoder: Rename virtual accessors to CamelCase by kwiberg@webrtc.org · 10 years ago
- cc483b7 Roll chromium_revision 601e6f3..b0c3ed3 (315263:316737) by kjellander@webrtc.org · 10 years ago
- b4987bf Send black frame with previous size when muting. by pbos@webrtc.org · 10 years ago
- 7d721ee Adding speech_expand_rate to NetEQ Network Statistics. by minyue@webrtc.org · 10 years ago
- 3864363 cricket::VideoFrame: Refactor CopyToBuffer into base class by magjed@webrtc.org · 10 years ago
- dd4a8da Remove DISABLE_YUV flag by magjed@webrtc.org · 10 years ago
- 97aaf68 Bump to version 42. by jansson@webrtc.org · 10 years ago
- bfa3c72 Don't call g_thread_init on glib >=2.31.0 by decurtis@webrtc.org · 10 years ago
- e9facf8 Add range checks in a variety of places where the values will subsequently be by pkasting@chromium.org · 10 years ago
- 27669f3 Apply good settings to Beamformer by aluebs@webrtc.org · 10 years ago
- b08f404 Fix issue 4061. by guoweis@webrtc.org · 10 years ago
- 0abc601 Remove SetCaptureDelay from the RTP module. by mflodman@webrtc.org · 10 years ago
- 7663684 Implement the Nada rmcat proposal within the simulation framework. by stefan@webrtc.org · 10 years ago
- 71b35a4 iLBC: Use uint8_t[] for byte arrays by jmarusic@webrtc.org · 10 years ago
- 640313c WebRtcVideoCapturer: Remove dead code |OnIncomingCapturedEncodedFrame| by magjed@webrtc.org · 10 years ago
- 7a91acb ViECapturer: Remove unimplemented function declaration |DeliverCodedFrame| by magjed@webrtc.org · 10 years ago
- 1a38a51 Add default implementation to VideoSourceInterface of Stop and Restart. by perkj@webrtc.org · 10 years ago
- a28a91d Fix data race for RTCPReceiver stats callback. by pbos@webrtc.org · 10 years ago
- 8f605e8 Add VideoSource::Stop and Restart methods. by perkj@webrtc.org · 10 years ago
- 959dac7 VideoCaptureImpl: Remove unused member variable |_capture_encoded_frame| by magjed@webrtc.org · 10 years ago
- 4dd40d6 Signal threads for faster receiver destruction. by pbos@webrtc.org · 10 years ago
- 0a7d4ee Remove usage of BitrateController in VoiceEngine. by mflodman@webrtc.org · 10 years ago
- f9b5c1b Removing CELT. by minyue@webrtc.org · 10 years ago
- 2c1bcf2 Adding decoded_fec_rate to NetEQ Network Statistics. by minyue@webrtc.org · 10 years ago
- 290cb56 Remove TimeToSendPacket and TimeToSendPadding from the default module. by mflodman@webrtc.org · 10 years ago
- c0fc4dd Add 'mac_x64' trybot to default set. by kjellander@webrtc.org · 10 years ago
- 86196c4 Setup encoders inexpensively before first frame. by pbos@webrtc.org · 10 years ago
- 34509d9 Fix an issue with comfort noise in ACMGenericCodecWrapper by henrik.lundin@webrtc.org · 10 years ago
- e9f0f59 Enable bitrate probing by default in PacedSender. by stefan@webrtc.org · 10 years ago
- fbc347f Re-land r8342 "Switch to using AudioEncoderIsac instead of ACMISAC"" by henrik.lundin@webrtc.org · 10 years ago
- ce22f13 GN: Changes for vp9, opus and direct trace by kjellander@webrtc.org · 10 years ago
- e35fa96 Move isacfix.gypi and isac.gypi by kjellander@webrtc.org · 10 years ago
- 0200f70 Set webrtc_rtp category to be disabled by default. by sprang@webrtc.org · 10 years ago
- 14b0279 Break out code from bloated files in the BWE simulator. by stefan@webrtc.org · 10 years ago
- 0f7f161 Add audio_coding module OWNERS file. by kjellander@webrtc.org · 10 years ago
- 4dc0003 Revert r8342 "Switch to using AudioEncoderIsac instead of ACMISAC" by henrik.lundin@webrtc.org · 10 years ago
- 30142bb Add arraysize to overrides to avoid macros redefinitions in Chromium by aluebs@webrtc.org · 10 years ago
- d3b453b Remove the incremental IP address behavior from virtualsocketserver by guoweis@webrtc.org · 10 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 · 10 years ago
- 92a19bc Simplify mask calculation by aluebs@webrtc.org · 10 years ago
- 56cb0ea Add support for bi-directional simulations by having both an uplink and a downlink. by stefan@webrtc.org · 10 years ago
- d5ce2e6 Remove EventWrapper::Reset(). by pbos@webrtc.org · 10 years ago
- 5a7dc39 This is a code clean up. No functional change intended. by guoweis@webrtc.org · 10 years ago
- a8cc344 Allowing RED decoding for Opus. by minyue@webrtc.org · 10 years ago