Commit Graph

  • 971eb27e54 Removing, opening and creating files in platform_file and file Viktor Palmkvist 2016-09-16 10:19:23 +02:00
  • 280ad1514e Update test code to use I420Buffer when writing pixel data. Niels Möller 2016-09-16 10:07:06 +02:00
  • e9a3c7f43c Wrap ScreenCapturer with ScreenCapturerDifferWrapper This change is to add an DesktopCapturerOptions accurate_updated_region() with default value as false to indicate whether a pixel-wise differentiation is required. And ScreenCapturer::Create() function will wrap the implementation with ScreenCapturerDifferWrapper. zijiehe 2016-09-16 00:03:15 -07:00
  • 2dc4cdeea7 Correcting the enabling of the level controller in the audio processing simulator peah 2016-09-16 00:01:35 -07:00
  • 74e1a4f96a PeerConnection[Interface]::GetStats(RTCStatsCollectorCallback*) added. hbos 2016-09-15 23:33:01 -07:00
  • fe7d091f57 Fixing a couple cases that cause ProcessAllMessageQueues to hang. Taylor Brandstetter 2016-09-15 17:47:42 -07:00
  • 9ecb08576e Adding logs to track potential cause of not starting port allocation. Honghai Zhang 2016-09-15 16:41:04 -07:00
  • fb2c1d0636 Add voe_cmd_test to voice_engine/BUILD.gn (and remove it from voice_engine.gyp, together with the channel_transport gyp target) solenberg 2016-09-15 13:12:02 -07:00
  • 232c56ba9a Add logging available fps ranges to Camera2Session. sakal 2016-09-15 12:31:17 -07:00
  • bb716daea7 Fix android_junit_tests and add a GN target for them. sakal 2016-09-15 12:27:52 -07:00
  • 530b3f5d06 Merge RtcpReceiver::Handle<Packet>Item functions into Handle<Packet> As a preparation to replace parsing implementation. Danil Chapovalov 2016-09-15 18:41:02 +02:00
  • 9fdbda6aa3 Revert of Replace interface VideoCapturerInput with VideoSinkInterface. (patchset #13 id:280001 of https://codereview.webrtc.org/2257413002/ ) perkj 2016-09-15 09:19:20 -07:00
  • 95a226f55a Replace VideoCapturerInput with VideoSinkInterface. Adds new method VideoSendStream::SetSource(rtc::VideoSourceInterface* and VieEncoder::SetSource(rtc::VideoSourceInterface*) perkj 2016-09-15 08:57:21 -07:00
  • 91511f13e1 Split RtcpReceiver::HandleSenderReceiverReport into two functions as a preparation to replace parsing implementation Danil Chapovalov 2016-09-15 16:24:02 +02:00
  • edebf45712 Use I420Buffer rather than VideoFrameBuffer when writing pixels. nisse 2016-09-15 07:20:40 -07:00
  • 8faf9e047e Removed the const char* (StaticString) type from RTCStatsMember. hbos 2016-09-15 06:52:43 -07:00
  • 4ed5b9f62d Android SurfaceViewRenderer: Create EGL context on render thread Magnus Jedvert 2016-09-15 15:30:17 +02:00
  • ec62374ccd Reland of Add arraysize to rtc_base_approved. Remove dependency of audio_device on rtc_base. (patchset #1 id:1 of https://codereview.webrtc.org/2340253003/ ) maxmorin 2016-09-15 05:11:55 -07:00
  • 0a6e0dc471 Disable all screen-capturer tests stefan 2016-09-15 05:04:34 -07:00
  • 17f008bf33 GYP: Remove targets inside include_tests==1 that are converted to GN. kjellander 2016-09-15 04:57:37 -07:00
  • 8b28b8017f Assume ProjectRootPath() equals ../.. in Desktop ehmaldonado 2016-09-15 04:45:48 -07:00
  • d268d6ffbe Stash non layer-sync frames if we have not yet received an earlier frame for this layer. philipel 2016-09-15 13:43:13 +02:00
  • ebc34e78db [GN] Add rtc_sdk_framework_objc target to GN kthelgason 2016-09-15 04:30:18 -07:00
  • 11ace15c19 The VoE functionality to apply receive-side processing to VoE channels is unused. I'm removing it so we can avoid instantiating a full APM per channel (and thus also for webrtc::AudioSendStream and webrtc::AudioReceiveStream), and then never use it. solenberg 2016-09-15 04:29:13 -07:00
  • 70d01242f8 MB: Change Android Clang to build shared instead of static. kjellander 2016-09-15 03:42:29 -07:00
  • 5a20ed36e6 Fix undefined reference to log2 on android Kári Tristan Helgason 2016-09-15 10:56:19 +02:00
  • 89fb9201b7 Revert of Add arraysize to rtc_base_approved. Remove dependency of audio_device on rtc_base. (patchset #1 id:1 of https://codereview.webrtc.org/2346763002/ ) maxmorin 2016-09-15 01:45:25 -07:00
  • 100c9d0266 Add arraysize to rtc_base_approved. Remove dependency of audio_device on rtc_base. maxmorin 2016-09-15 01:40:35 -07:00
  • 705ecc5dda GN: Change group deps to public_deps. kjellander 2016-09-15 00:53:26 -07:00
  • c26f77f5a6 Remove a couple of unnecessary dependencies on gflags henrik.lundin 2016-09-15 00:04:55 -07:00
  • f807a521fa iSAC: Remove unnecessary WEBRTC_LINUX define kjellander 2016-09-14 23:15:48 -07:00
  • 0d14c6abba Add method cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource. nisse 2016-09-14 12:03:17 -07:00
  • 8295c9326d [WebRTC] Add TwoCapturers test and TwoMagnifierCapturers test zijiehe 2016-09-14 10:21:57 -07:00
  • 3626d7e247 Move CopyOnWriteBuffer functions definitions from .h to .cc Danil Chapovalov 2016-09-14 17:14:30 +02:00
  • 2e164c6b53 Adding ChannelController to audio network adaptor. minyue 2016-09-14 06:47:36 -07:00
  • fdafab84bc Fix issues with rtc_stats_unittests tests so that they can run on bots. This target is not run on bots so a couple of issues went under the radar. If we expose the tests and run them on the bots[1] two issues are surfaced which this CL fixes. After this CL lands we can enable this target on the bots without it going red. hbos 2016-09-14 06:02:13 -07:00
  • 6fa69c91d6 Relaxed unnecessarily stringent thread checking in WebRtcAudioSendStream::OnData(). solenberg 2016-09-14 06:01:32 -07:00
  • cbae0b475c Use I420Buffer rather than VideoFrameBuffer when writing pixels. nisse 2016-09-14 05:45:06 -07:00
  • bc18fc07be Change onCameraOpening to take camera name as a parameter instead of camera id. sakal 2016-09-14 05:36:20 -07:00
  • 9e2be5f292 webrtc/modules/audio_processing: Use RTC_DCHECK() instead of assert() kwiberg 2016-09-14 05:23:22 -07:00
  • 3a7f35b1c4 GN: Declare resources for targets. ehmaldonado 2016-09-14 05:10:01 -07:00
  • 52a5703721 Enable BWE logging to command line when rtc_enable_bwe_test_logging is set to true gaetano.carlucci 2016-09-14 05:04:36 -07:00
  • b471d1cffb Android EglBase: Include EGL error code in exceptions magjed 2016-09-14 02:40:50 -07:00
  • 194f40a2e7 Refactor QualityScaler and MovingAverage kthelgason 2016-09-14 02:14:58 -07:00
  • a075848ebd New method TimestampAligner::TranslateTimestamp nisse 2016-09-14 00:37:00 -07:00
  • f8a4ecc4a1 Remove dependency of audio_device on metrics_default. maxmorin 2016-09-14 00:20:22 -07:00
  • 17366bc090 Remove handling unused rtcp packets. App, ExtendedJitterReport and VoipMetric in ExtenedReports are not used when received (no callbacks, no state change), so removed. danilchap 2016-09-13 23:54:48 -07:00
  • cdf37a9293 Delete Timing class, timing.h, and update all users. nisse 2016-09-13 23:41:47 -07:00
  • d29e3ea4b2 Added build flag around the Intelligibility enhancer performance test code peah 2016-09-13 21:42:39 -07:00
  • caa9cb2cea Adding basic implementation of AudioNetworkAdaptor. minyue 2016-09-13 13:34:15 -07:00
  • dd12892ede Reland of Make rtcp parsing implementation private in RtcpReceiver (patchset #1 id:1 of https://codereview.webrtc.org/2332673003/ ) danilchap 2016-09-13 12:23:29 -07:00
  • d59d3bb117 Replace a DCHECK with static_assert kwiberg 2016-09-13 07:49:33 -07:00
  • ba56b6c7d2 Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ solenberg 2016-09-13 07:29:12 -07:00
  • bb723e53b4 Fixed video_loopback target. charujain 2016-09-13 05:52:48 -07:00
  • 2b2779f3f1 Make CopyOnWriteBuffer keep capacity for SetData and Clear functions too. Danil Chapovalov 2016-09-13 14:15:10 +02:00
  • 9708884c31 Update rtcp receiver fuzzer to use generic function IncomingPacket(const uint8_t* packet, size_t size) instead of implementation specific IncomingRTCPPacket(PacketInfo* out, Parser* in) This would allow switch parse implementation Danil Chapovalov 2016-09-13 12:41:41 +02:00
  • 6631e8a21b Minor fixes in FEC and RtpSender{,Video} brandtr 2016-09-13 03:23:29 -07:00
  • 07d9e545ff Revert of Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ (patchset #7 id:120001 of https://codereview.webrtc.org/2319583005/ ) solenberg 2016-09-13 01:24:03 -07:00
  • 22487b2d0a webrtc/base: Use RTC_DCHECK() instead of assert() kwiberg 2016-09-13 01:17:10 -07:00
  • ade2a038a9 Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ solenberg 2016-09-13 01:10:46 -07:00
  • 88ac853e14 The current scheme for setting parameters and specifying the behavior of the audio processing module is quite complex and hard to implement in a threadsafe and efficient manner. Therefore a new scheme for setting the parameters in the audio processing module is introduced in this CL. peah 2016-09-12 16:47:25 -07:00
  • b2540bb99f Probing: Add support for exponential startup probing Irfan Sheriff 2016-09-12 12:28:54 -07:00
  • a421ddd60e The buffering of the farend signal is refactored in this CL. peah 2016-09-12 11:27:14 -07:00
  • b3f7876a95 Add printStackTrace method to CameraCapturer. sakal 2016-09-12 07:35:18 -07:00
  • 78ce619a0c Extract simulcast rate allocation outside of video encoder. Erik Språng 2016-09-12 16:04:43 +02:00
  • 7b11c65a17 MB: Move iOS GYP bots to use limited support config kjellander 2016-09-12 05:33:54 -07:00
  • 8e56521143 The output signal of the AEC needs to be buffered as the internal block size of the AEC differ from the frame size in the AEC output. peah 2016-09-12 04:49:45 -07:00
  • a64a2fbf6d Fix oversized rtp extension parsing. When size of individual one-byte extension span beyound extension block Danil Chapovalov 2016-09-12 11:41:35 +02:00
  • 180e4525ca Revert of Make rtcp parsing implementation private in RtcpReceiver (patchset #1 id:1 of https://codereview.webrtc.org/2320603002/ ) danilchap 2016-09-12 01:40:40 -07:00
  • faf708e238 Make rtcp parsing implementation private in RtcpReceiver: Function just for Parse and for Callbacks moved to private section. All handles moved from protected to private section. Danil Chapovalov 2016-09-12 10:31:11 +02:00
  • 1a0533dc3d Add statistics for the time it takes to start and stop the camera on Camera2. sakal 2016-09-12 01:05:25 -07:00
  • 6ffb67d049 Add periodic logging of number of captured and dropped frames in VideoCaptureInput. Logged every minute. asapersson 2016-09-12 00:10:45 -07:00
  • 11d5766819 GN: Revert to default compiler optimizations for Win Release. kjellander 2016-09-11 23:29:36 -07:00
  • 10f606d8de Revert of Introduced new scheme for controlling the functionality inside the audio processing module (patchset #12 id:260001 of https://codereview.webrtc.org/2292863002/ ) kjellander 2016-09-11 23:04:31 -07:00
  • 5df5434cef Fix a type mistake honghaiz 2016-09-10 15:47:52 -07:00
  • 2ace3f9406 The audio processing module (APM) relies on two for functionalities doing sample-rate conversions: -The implicit resampling done in the AudioBuffer CopyTo, CopyFrom, InterleaveTo and DeinterleaveFrom methods. -The multi-band splitting scheme. peah 2016-09-10 04:42:27 -07:00
  • 1d02d3e5e6 Remove RTC_LOGGED_* macro. asapersson 2016-09-09 22:40:25 -07:00
  • d5fff5040c Removing assert error when we fail to create a connection for a ping from an unknown address. Honghai Zhang 2016-09-09 20:47:53 -07:00
  • ed0b0dba15 Revert "Optimize Android NV12 capture" jackychen 2016-09-09 16:15:11 -07:00
  • c8bbe3fe9a The current scheme for setting parameters and specifying the behavior of the audio processing module is quite complex and hard to implement in a threadsafe and efficient manner. Therefore a new scheme for setting the parameters in the audio processing module is introduced in this CL. peah 2016-09-09 14:15:57 -07:00
  • e753641ef1 Adding ability to simulate EWOULDBLOCK/SignalReadyToSend. Taylor Brandstetter 2016-09-09 13:16:15 -07:00
  • fc433e6546 Don't use VoE legacy APIs in force_mic_volume_max tool. solenberg 2016-09-09 11:04:53 -07:00
  • fac0ff022b Change SimulcastEncoderAdapter to allow a 0 for SetRates. noahric 2016-09-09 10:27:15 -07:00
  • 36d38cbb15 Optimize Android NV12 capture magjed 2016-09-09 09:09:46 -07:00
  • 291cd8fac3 CopyOnWriteBuffer::SetSize to smaller size memcpy less. Danil Chapovalov 2016-09-09 17:32:33 +02:00
  • 96f2c4df31 Remove unused audio_e2e_harness.cc and infrastructure. solenberg 2016-09-09 06:51:18 -07:00
  • 467bc84dc9 Revert webrtc/build/mb_config.pyl accidental change Henrik Kjellander 2016-09-09 15:07:15 +02:00
  • a41c13e6a2 OWNERS: Make everyone able to change *.gn,*.gni files. Henrik Kjellander 2016-09-09 14:51:37 +02:00
  • 2b1b7a83ad iSAC fix: Ignore overflow in signed left shift kwiberg 2016-09-09 05:51:33 -07:00
  • 53cec04f5c GN: Move audio_coding to public_deps in voice engine ehmaldonado 2016-09-09 05:32:14 -07:00
  • f06f35a161 Adds logging of native audio levels and UMA stats to track issues. henrika 2016-09-09 14:23:11 +02:00
  • 99f8e0841a Add a chart for packet loss on incoming streams. Stefan Holmer 2016-09-09 13:37:50 +02:00
  • 073378e79a Avoids crash at device switch on iOS by ensuring that audio parameters can be updated on the fly driven by e.g. switching audio device. henrika 2016-09-09 13:15:37 +02:00
  • 2b11fd2e4b rtc::Optional: Tell sanitizers that unset values aren't OK to access kwiberg 2016-09-09 03:35:24 -07:00
  • 463d3011ce Added ClearTo(seq_num) to RtpFrameReferenceFinder. philipel 2016-09-09 03:32:44 -07:00
  • d5472246a5 Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ ) kthelgason 2016-09-09 03:19:48 -07:00
  • 27c7b8ff11 VadCore: Allow signed multiplication overflow that we don't know how to fix kwiberg 2016-09-09 02:04:38 -07:00
  • 3fa35172cd Filter objc headers in cpplint presubmit check Kári Tristan Helgason 2016-09-09 08:55:05 +00:00
  • 9c8c586dd3 MB: Disable more parts of the GYP build. kjellander 2016-09-09 01:42:06 -07:00
  • 499dcb1ac0 Remove references to .isolate files that are no longer needed. kjellander 2016-09-09 01:38:52 -07:00