generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Commit Graph

  • f353dd59b5 VoE: cleanup VoENetwork implementation Jelena Marusic 2015-05-06 15:04:22 +02:00
  • 1ff218fac3 audio_processing/aec: Do not scale target delay at startup when on Android Bjorn Volcker 2015-05-06 12:08:38 +02:00
  • 532531b656 audio_processing/delay_estimator: Always update robust validation statistics Bjorn Volcker 2015-05-06 11:58:04 +02:00
  • 40a6d593d2 audio_processing/tests: Adds a flag to unpack input data to text file Bjorn Volcker 2015-05-06 10:51:34 +02:00
  • 9695d8523b Added VP9FrameBufferPool, a memory pool that is shared between libvpx and webrtc. Using the VP9 codec, the libvpx decoder will obtain its buffers from our memory pool. This lets us reuse the same buffers for our I420VideoFrames and not have to copy a frame for every decode (from libvpx buffers to webrtc/I420VideoFrame buffers). Henrik Boström 2015-05-06 10:42:15 +02:00
  • f242e665b4 Replace asm NEON function by intrinsics implementation on ARMv7 Zhongwei Yao 2015-05-06 16:39:17 +08:00
  • 507a550af8 Delete auto-roll script since moved into Chromium. Henrik Kjellander 2015-05-06 10:32:47 +02:00
  • 589699eea2 Fix bug in transform_neon.c in iSAC codec. Zhongwei Yao 2015-05-06 10:25:04 +08:00
  • 57cc74e32c iOS camera switching video capturer. Zeke Chin 2015-05-05 07:52:31 -07:00
  • 5cb9ce4c74 Remove ViECodec usage in VideoSendStream. Peter Boström 2015-05-05 15:16:30 +02:00
  • ab00404571 VCMEncodedFrame::VerifyAndAllocate: Use size_t instead of uint32_t for size argument Magnus Jedvert 2015-05-05 11:37:12 +02:00
  • 01b488831b Use padding to achieve bitrate probing if the initial key frame has too few packets. Stefan Holmer 2015-05-05 10:21:24 +02:00
  • 78c8bbfa34 Roll chromium_revision 0cb2549..ec5b768 (327252:328242) Henrik Kjellander 2015-05-05 09:55:10 +02:00
  • c56ac1ec29 rtc::Buffer: Remove backwards compatibility band-aids Karl Wiberg 2015-05-04 14:54:55 +02:00
  • f75f0cf36a Enable GoogleWifiTrace3Mbps simulations. Stefan Holmer 2015-05-04 14:26:20 +02:00
  • 0d266054ac VoE: apply new style guide on VoE interfaces and their implementations Jelena Marusic 2015-05-04 14:15:32 +02:00
  • 79c143312b Delete VoiceChannelTransport before deleting Channel in voe_cmd_test Minyue Li 2015-05-04 11:21:00 +02:00
  • 0b15445fd5 VoE: Follow-up to https://webrtc-codereview.appspot.com/49759004/ Jelena Marusic 2015-05-04 09:55:59 +02:00
  • e433c0ef31 Restore back verbosity logging for camera captured frame. Alex Glaznev 2015-05-01 13:54:19 -07:00
  • f2f828374c Use rtc::CriticalSection in webrtc/video/. Peter Boström 2015-05-01 13:00:41 +02:00
  • cac1b38135 Expose RTCConfiguration to java JNI and add an option to disable TCP Jiayang Liu 2015-04-30 12:35:24 -07:00
  • 4eddf18b1c Don't crash if SetRemoteDescription is called first with BundlePolicy=max-bundle. Peter Thatcher 2015-04-30 10:55:59 -07:00
  • 8a6680e9ec Remove base/move.h (no one uses it anymore) Karl Wiberg 2015-04-30 16:06:17 +02:00
  • cbf0927473 Revert "rtc::Buffer: Remove backwards compatibility band-aids" Karl Wiberg 2015-04-30 16:00:56 +02:00
  • 9e1a6d7c23 rtc::Buffer: Remove backwards compatibility band-aids Karl Wiberg 2015-04-30 14:25:03 +02:00
  • ff019b0b55 Move rtc::AtomicOps to webrtc/base/atomicops.h. Peter Boström 2015-04-30 14:16:07 +02:00
  • f16fcbec73 Remove ViECapture usage in VideoSendStream. Peter Boström 2015-04-30 12:16:05 +02:00
  • 46bd31b994 VoE: VoENetwork unit test Jelena Marusic 2015-04-30 10:57:10 +02:00
  • 3cfa756f37 audio_processing/aec: Fixes an incorrect sampling rate multiplier when processing in 48 kHz Bjorn Volcker 2015-04-29 20:22:44 +02:00
  • efbde3775b Don't use CPU adaptation for screen content in the new API. Erik Språng 2015-04-29 16:21:28 +02:00
  • adf89b7e33 Added SetBitRate function to VoE API to allow changing the audio bitrate. Ivo Creusen 2015-04-29 16:03:33 +02:00
  • 23fba1ffa0 Add AudioReceiveStream to Call API. Fredrik Solenberg 2015-04-29 15:24:01 +02:00
  • 10ba3eec5a Roll chromium_revision a12e1e1..0cb2549 (326495:327252) Henrik Kjellander 2015-04-29 14:47:53 +02:00
  • dea11f9c43 Add per flow throughput and delay metrics. Stefan Holmer 2015-04-29 14:27:35 +02:00
  • 94cc1fe4af Remove ViEImageProcess usage in VideoSendStream. Peter Boström 2015-04-29 14:08:41 +02:00
  • c444de6276 Make setup_links.py handle non-link directories during cleanup Henrik Kjellander 2015-04-29 11:27:22 +02:00
  • 1ba344a070 Adds a MediaConstraint for the AudioOption aec_dump Bjorn Volcker 2015-04-29 07:28:10 +02:00
  • 97f13c5f7f Fixed incorrect RBSP parsing. The original version would eat 0x3 as an emulation byte in places where it shouldn't, whereas the real parsing is only supposed to eat 0x3 preceded by 0x0 0x0. Noah Richards 2015-04-28 17:55:36 -07:00
  • 86153c26a0 Added a BitBufferWriter subclass that contains methods for writing bit and byte-sized data, along with exponential golomb encoded data. Noah Richards 2015-04-28 15:13:44 -07:00
  • 80154f6b28 Set correct .type directive for asm functions. Wei Zhong 2015-04-28 13:51:53 -07:00
  • faa6d076b7 Remove a few verbose log messages from webrtcvideoengine2. Alex Glaznev 2015-04-28 09:40:39 -07:00
  • 019087f5bb Add safeguards against signalling peer-reflexive candidates. Peter Thatcher 2015-04-28 09:06:26 -07:00
  • ae331349c6 Always specify current OS when syncing Chromium. Henrik Kjellander 2015-04-28 16:08:23 +02:00
  • 8786f637b2 Roll gtest-parallel. Peter Boström 2015-04-28 15:36:09 +02:00
  • 31dc737d7a Platform dependent way of generating the seed for srand for simulations, so that they can be run in parallel. Stefan Holmer 2015-04-28 15:32:33 +02:00
  • 88de4792d0 AudioEncoderIsac: Print error code if CHECK for successful encoding fails Karl Wiberg 2015-04-28 14:58:43 +02:00
  • bcbcd84888 Improve TCP implementation by adding ssthresh and make it possible to start it with an offset. Stefan Holmer 2015-04-28 14:39:00 +02:00
  • 9d657cfd66 Fix dangling pointer in screenshare_loopback Erik Språng 2015-04-28 14:00:58 +02:00
  • beb9798ab4 audio_processing: Fixed incorrect usage of SetExtraOptions() in offline tool Bjorn Volcker 2015-04-28 13:52:50 +02:00
  • ddbddbdee6 Remove ViENetwork usage in VideoSendStream. Peter Boström 2015-04-28 12:35:44 +02:00
  • 038df3c5d7 Remove ViEExternalCodec usage in VideoSendStream. Peter Boström 2015-04-28 11:58:44 +02:00
  • 4a9cb6b67d Prevent zero-timestamps in captured_frame_. Peter Boström 2015-04-28 10:48:43 +02:00
  • 143cec1cc6 Set correct encoder-specific settings for vpx in the new API. Erik Språng 2015-04-28 10:01:41 +02:00
  • e8a197bd07 Enable isac NEON building on Aarch64 Zhongwei Yao 2015-04-28 14:42:11 +08:00
  • d7e5c44e94 STUN allocation should not be disabled when using shared port and TURN servers are provided. Jiayang Liu 2015-04-27 11:47:21 -07:00
  • 5a92aa8440 Add 3-band filter-bank implementation Alejandro Luebs 2015-04-27 11:34:45 -07:00
  • 494f20977e Move CriticalSection into rtc_base_approved. Tommi 2015-04-27 17:39:23 +02:00
  • 59d91dc951 Remove ViERTP_RTCP usage in VideoSendStream. Peter Boström 2015-04-27 17:24:33 +02:00
  • e6cefb60f8 GYP variables for building expat, icu, libsrtp, usrsctp Henrik Kjellander 2015-04-27 14:39:04 +02:00
  • 61be2a4016 Clean up RTCPSender. Erik Språng 2015-04-27 13:32:52 +02:00
  • 3c391cbabb Add support for updating histogram for received fraction loss ("WebRTC.Video.ReceivedPacketsLostInPercent") when running new video api. Åsa Persson 2015-04-27 10:09:49 +02:00
  • 52ef9d7738 Stop IncomingVideoStream on delete. Peter Boström 2015-04-24 18:06:52 +02:00
  • 23dc68e515 Add the rtc_build_openmax_dl variable to the GN build. Andrew MacDonald 2015-04-24 08:46:51 -07:00
  • 12e0329007 Do not use Magnifier if there are multiple screens since it sometimes crashes. Jiayang Liu 2015-04-24 08:46:35 -07:00
  • 77d444a433 Handle the case when hoststring is empty. Tommi 2015-04-24 15:38:38 +02:00
  • c4188fd3c7 Use IncomingVideoStream in VideoReceiveStream. Peter Boström 2015-04-24 15:16:03 +02:00
  • f955b5d3f5 Add h.264 AVC SPS parsing for resolution (re-land) Henrik Kjellander 2015-04-24 13:57:06 +02:00
  • c043afc605 Cleanup inside IncomingVideoStream. Peter Boström 2015-04-24 13:54:27 +02:00
  • a9ae0dfe12 Roll chromium_revision d5098d0..a12e1e1 (326014:326495) Henrik Kjellander 2015-04-24 09:27:55 +02:00
  • a96f02b6f3 Make sure histograms in jitter buffer are only updated if running. Åsa Persson 2015-04-24 08:52:11 +02:00
  • affcfb2f16 Refactor common_audio/signal_processing: Removed usage of trivial macro WEBRTC_SPL_MUL_16_16 Bjorn Volcker 2015-04-24 08:12:07 +02:00
  • e3827f27c3 Revert "Add h.264 AVC SPS parsing for resolution." Noah Richards 2015-04-23 18:15:19 -07:00
  • 5ea8eff55e Add h.264 AVC SPS parsing for resolution. Noah Richards 2015-04-23 16:45:56 -07:00
  • 9728241e6a Record H264 NALU type in the h264 header. Noah Richards 2015-04-23 11:15:08 -07:00
  • fe7a80c38c Prevent sender RTCP signals for receive-only channels. Peter Boström 2015-04-23 17:53:17 +02:00
  • 7f287cca67 rtc::CriticalSection: Add dummy implementation of IsLocked for release builds Magnus Jedvert 2015-04-23 16:06:59 +02:00
  • 24d4485614 Enable -Wunused-private-field warning for talk/ Henrik Kjellander 2015-04-23 14:51:18 +02:00
  • d3e8eda839 (Re-land) AudioEncoderDecoderIsac: Merge the two config structs Karl Wiberg 2015-04-23 14:07:06 +02:00
  • 92f9eacd13 g722 and red encoders: Use rtc::Buffer instead of scoped_ptr<uint8_t[]> Karl Wiberg 2015-04-23 13:53:22 +02:00
  • 261f644ce3 Suppressing VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly on Dr.Memory Minyue Li 2015-04-23 13:46:35 +02:00
  • 6bf10843bf rtc::CriticalSection: Add function IsLocked Magnus Jedvert 2015-04-23 11:37:55 +02:00
  • bd67f66ebd Restore webrtc/base/move.h, because it's used in Windows Chromium builds Karl Wiberg 2015-04-23 09:52:33 +02:00
  • 352595459d Use short include paths for icu headers. Henrik Kjellander 2015-04-23 08:58:21 +02:00
  • 915590e41f Moved ByteBuffer/BitBuffer into rtc_base_approved. Noah Richards 2015-04-22 15:43:08 -07:00
  • 01aeaee719 Fix GetSignatureDigestAlgorithm for openssl to prepare for EC key switch. JiaYang (佳扬) Liu 2015-04-22 12:18:33 -07:00
  • a8e285d193 Remove webrtc/base/move.h, and make types move-only manually Karl Wiberg 2015-04-22 19:44:19 +02:00
  • ee0b00e8a9 Prevent recv-stream reconfig on identical codecs. Peter Boström 2015-04-22 18:41:14 +02:00
  • 908e77bd00 Allow Java code to detect if VP8 and H.264 HW decoding is supported. Alex Glaznev 2015-04-22 09:25:34 -07:00
  • b67288283a Move cricket::FakeCall and associates to a separate file. Fredrik Solenberg 2015-04-22 15:35:17 +02:00
  • 7fb711f683 Remove unused voice channel argument from cricket::VideoChannel ctor and corresponding field in class. Fredrik Solenberg 2015-04-22 15:30:51 +02:00
  • 96d1d89c3e Do not register bandwidth observer for receive only channels. An incoming rtcp report block is inserted to both send and receive channels in Call::DeliverRtcp. The report block may also be accepted by each receive channel (in addition to the send channel) but fails to calculate the rtt (=0). Remove registration of bandwidth observer for receive channels. Prevents multiple callbacks to the bitrate controller (and with incorrect rtt) for an incoming report block. Åsa Persson 2015-04-22 14:57:50 +02:00
  • 393347ff98 Report receive-side packet loss. Peter Boström 2015-04-22 14:52:45 +02:00
  • 7c027b64ae Enable more Clang warnings for talk/ Henrik Kjellander 2015-04-22 13:21:30 +02:00
  • 5a3178042b Reformatting RTPtimeshift.cc file. Ivo Creusen 2015-04-22 13:12:00 +02:00
  • ac69016b0f Improve TCP by adding a real timeout to in flight packets. Stefan Holmer 2015-04-22 13:11:42 +02:00
  • 8e4b9e8804 Roll chromium_revision dcb0929..d5098d0 (325030:326014) Henrik Kjellander 2015-04-22 08:50:23 +02:00
  • e555b7b440 Fix CC flags in GN Windows build. Henrik Kjellander 2015-04-22 08:49:52 +02:00
  • fb49451014 Disables mic bump-up level if not built with chromium Bjorn Volcker 2015-04-22 06:39:58 +02:00
  • 8f85dbcce4 Reduce the number of registers used in MIPS optimizations. Ljubomir Papuga 2015-04-21 16:52:45 -07:00
  • bbf7c864ad Add a new BitBuffer class to webrtc base. Noah Richards 2015-04-21 16:30:13 -07:00