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2014-11-06Increase speed setting for VP9 (from 5 to 6) and re-enable end_to_end test.marpan@webrtc.org
TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28949004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7637 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05Wire up bandwidth stats to the new API and webrtcvideoengine2.stefan@webrtc.org
Adds stats to verify bandwidth and pacer stats. BUG=1788 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24969004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Add support for VP9 in webrtc::Call and video_loopback.stefan@webrtc.org
R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31959004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7622 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Adds support for finch experiments to video_loopback.stefan@webrtc.org
Adds support for logging to stderr via -logs. Enables abs-send-time by default. R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23329004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7613 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Delete VideoReceiveStream channels in destructor.pbos@webrtc.org
R=stefan@webrtc.org BUG=1667 Review URL: https://webrtc-codereview.appspot.com/31909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01Add VP9 codec to VCM and vie_auto_test.marpan@webrtc.org
Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Implement conference-mode temporal-layer screencast.pbos@webrtc.org
Renames VideoStream::temporal_layers to temporal_layer_thresholds_bps to convey that it contains thresholds needed to ramp up between them (1 threshold -> 2 temporal layers, etc.). R=mflodman@webrtc.org, stefan@webrtc.org BUG=1788,1667 Review URL: https://webrtc-codereview.appspot.com/23269004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7578 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Configure A/V sync in WebRtcVideoEngine2.pbos@webrtc.org
Sets up A/V sync for the first video receive channel with the default voice channel. This is only done when conference mode is disabled to preserve existing behavior. Ideally we'd know which voice channel to sync with here. R=mflodman@webrtc.org, stefan@webrtc.org BUG=1788 Review URL: https://webrtc-codereview.appspot.com/23249004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7577 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29Use external VideoDecoders in VideoReceiveStream.pbos@webrtc.org
Removes direct VideoCodec use from the new API, exposes VideoDecoders through webrtc/video_decoder.h similar to VideoEncoders. Also includes some preparation for wiring up external decoders in WebRtcVideoEngine2 by adding AllocatedDecoders that specify whether they were allocated internally or externally. Additionally addresses a data race in VideoReceiver that was exposed with this change. R=mflodman@webrtc.org, stefan@webrtc.org TBR=pthatcher@webrtc.org BUG=2854,1667 Review URL: https://webrtc-codereview.appspot.com/27829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7560 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-24Move min transmit bitrate to VideoEncoderConfig.pbos@webrtc.org
min_transmit_bitrate_bps needs to be reconfigurable during a call (since this is currently set only for screensharing through libjingle and can't be set once and for all for the entire Call. R=mflodman@webrtc.org, stefan@webrtc.org BUG=1667 Review URL: https://webrtc-codereview.appspot.com/28779004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7518 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22Make ReconfigureVideoEncoder use current bitrate.pbos@webrtc.org
Prevents bitrate drops when changing resolution etc. R=stefan@webrtc.org BUG=1667 Review URL: https://webrtc-codereview.appspot.com/24069004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7493 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-20Make avg_{psnr,ssim}_threshold_ const.pbos@webrtc.org
Triggered warning on next clang version being rolled as these variables are annotated to be protected by crit_. R=stefan@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/24949004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7475 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."henrike@webrtc.org
BUG=3932 R=marpan@google.com Review URL: https://webrtc-codereview.appspot.com/27779004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7470 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14Remove -1 from Call::Config::start_bitrate_bps.pbos@webrtc.org
Instead initialize it to a good default value. The code does the same, but we don't have to check explicitly for -1. R=mflodman@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/23989004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7445 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14Add a packet loss full stack test to the new API.stefan@webrtc.org
Remove all full stack tests for the old API. BUG=3750 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23029004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7442 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10Disable SendsAndReceivesVP9 test for now.marpan@webrtc.org
Fails on linux memcheck and DrMemory. Will re-enable on next libvpx roll. TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27699004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7424 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10Add VP9 codec to VCM and vie_auto_test.marpan@webrtc.org
Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. Passes trybots. R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29449004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7422 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE. xians@webrtc.org
This also marks all virtual overrides of other classes in the same files. This will make a subsequent change I intend to do safer, where I'll change the argument types of the base Transport functions, by breaking the compile if I miss any overrides. This also highlighted a number of unused functions. I've removed some of these. TBR=mflodman@webrtc.org, pkasting@chromium.org BUG=none TEST=none Review URL: https://webrtc-codereview.appspot.com/28709004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-03Wire up CPU adaptation in WebRtcVideoEngine2.pbos@webrtc.org
Includes clean-up work to be able to use the webrtc::Call::Config that's set up. This introduced a CallFactory to spawn a FakeCall with the config used and allowed removal of FakeWebRtcVideoChannel2. BUG=1788 R=mflodman@webrtc.org, pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24779004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7370 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-28GN: Add common configs to all targets.kjellander@webrtc.org
This is needed to ensure we have the same build with GN as with GYP, since GYP includes the common.gypi on a global level. Several fixes has been needed in the past because some code have been built without the right defines. BUG=3441 R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/28589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7317 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Move thread_annotations.h to webrtc/base/.pbos@webrtc.org
R=andresp@webrtc.org, mflodman@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/27579004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22Set number of temporal layers for VideoSendStream.pbos@webrtc.org
Introduces a mapping between EncoderConfig and VideoCodec. More specifically it also removes an assert that there should be no set temporal layers in the new API, which is wrong and was temporary. R=stefan@webrtc.org BUG=1788 Review URL: https://webrtc-codereview.appspot.com/25619004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7256 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19Config struct for VideoEncoder.pbos@webrtc.org
Used for config parameters in common between multiple codecs as well as the encoder-specific pointer. In particular this contains content mode (realtime video vs. screenshare). BUG=1788 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16319004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7239 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19Re-enable missing android tests disabled due to issue 3770.andresp@webrtc.org
BUG=3770 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26519004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7238 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18Expose VP8/H264 defaults through video_encoder.h.pbos@webrtc.org
Reduces code duplication quite a bit, these identical defaults were set in quite a few different places. R=mflodman@webrtc.org, stefan@webrtc.org BUG=3070 Review URL: https://webrtc-codereview.appspot.com/19299004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7220 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18Split video_render_module implementation into default and internal ↵andresp@webrtc.org
implementation. Targets must now link with implementation of their choice instead of at "gyp"-time. Targets linking with libjingle_media: - internal implementation when build_with_chromium=0, default otherwise. Targets linking with default render implementation: - video_engine_tests - video_loopback - video_replay - anything dependent on webrtc_test_common Targets linking with internal render implementation: - vie_auto_test - video_render_tests - libwebrtcdemo-jni - video_engine_core_unittests GN changes: - Not many since there is almost no test definitions. Work-around for chromium: - Until chromium has updated libpeerconnection to link with video_capture_impl and video_render_impl, webrtc target automatically depends on it. This should fix the FYI bots and not require a webrtc roll to fix. Re-enable android tests by reverting 7026 (some tests left disabled). TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in. BUG=3770 R=kjellander@webrtc.org, pbos@webrtc.org TBR=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19359004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7217 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""pbos@webrtc.org
Re-lands r7114 after landing r7204 to adress the compile error causing the rollback in r7151. BUG=3070 TBR=henrikg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28489004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7207 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-15Re-enable rampup_tests.cc for Android.andresp@webrtc.org
BUG=3770 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27499004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7180 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-15Re-enable video send stream tests for android.andresp@webrtc.org
BUG=3770 R=kjellander@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7179 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."henrikg@webrtc.org
Speculative revert, seems to be reason for flaky Win FYI bot compile break. > Expose VideoEncoders with webrtc/video_encoder.h. > > Exposes VideoEncoders as part of the public API and provides a factory > method for creating them. > > BUG=3070 > R=mflodman@webrtc.org, stefan@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/21929004 TBR=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19329004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7151 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09Expose VideoEncoders with webrtc/video_encoder.h.pbos@webrtc.org
Exposes VideoEncoders as part of the public API and provides a factory method for creating them. BUG=3070 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21929004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7114 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04Change return value for number of discarded packets to be int.asapersson@webrtc.org
R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14209004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7054 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04Fix audio/video sync when FEC is enabled.stefan@webrtc.org
Also improves the tests by adding a test case for FEC, and running the a/v sync tests with NACK and simulated packet loss. BUG=crbug/374104 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19209004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7053 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03Network up/down signaling in Call.pbos@webrtc.org
BUG=2429 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13109005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7044 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02Disable video_engine_tests and webrtc_perf_tests on Android.kjellander@webrtc.org
BUG=3770 TESTED=Running the tests locally on an Android device. R=phoglund@webrtc.org TBR=henrik.lundin@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14299004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7026 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-29GN: Fix webrtc/video/BUILD.gn for Chromium build.kjellander@webrtc.org
A mistake was made in https://review.webrtc.org/18709004/ so it doesn't build in Chromium. Adding a config to get the root folder included in the include path solves it. BUG=3441 TESTED=Local compilation of Chromium's all target with src/third_party/webrtc linked to the WebRTC checkout with this CL applied. TBR=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/19169004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7011 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28GN: Implement video_engine, video_capture and video_render.kjellander@webrtc.org
Also add more from common.gypi to webrtc.gni. These GN configs are based on GYP files in r6997. BUG=3441 TEST=Trybots and local compile using: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default Passed compile from a Chromium checkout with src/third_party/webrtc linked to the webrtc/ dir of a checkout with this patch applied. R=brettw@chromium.org, glaznev@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18709004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6999 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26Disable EndToEndTest.RestartingSendStreamPreservesRtpState in ↵aluebs@webrtc.org
video_engine_tests because it is flaky BUG=webrtc:3745 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6981 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-14Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics().stefan@webrtc.org
Fixes issues where statistics only was reported for the first stream if configured with simulcast, and in case of RTX the reported statistics was depending on the order of the report blocks. Also fixes issues with multiple report blocks in the SendStatisticsProxy and the RtcpStatisticsCallback. SendStatisticsProxy is now aware of RTX ssrcs, and the RTCPReceiver is calling the RtcpStatisticsCallback with the correct SSRCs, and not only the primary stream SSRC. R=mflodman@webrtc.org, sprang@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20149004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6903 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-07Fix so video_replay logs aren't spammed.pbos@webrtc.org
Add unknown-SSRC counters instead and log number of unknown packets at end of session. R=stefan@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/13119004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6845 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-06RTP video playback tool using Call APIs.pbos@webrtc.org
Plays back rtpdump files from Wireshark in realtime as well as save the resulting raw video to file. Unlike the RTP playback tool it doesn't support faster-than-realtime playback/rendering, but it instead utilizes the same path as production code and also contains support for playing back FEC. BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16969004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6838 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-06Add end-to-end H.264 packetization test.stefan@webrtc.org
Also correctly wires up H.264 packetization in the new Call api. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20079004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6835 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-05Fix some code styles.pbos@webrtc.org
BUG= R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22009004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6830 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-31Add simulation of network effects to video_loopback tool.stefan@webrtc.org
Also add support for uniform random packet loss to the fake network pipe. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14039004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6803 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-24Make sure padding is sent on the first sending RTP module.mflodman@webrtc.org
R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13989004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6774 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-23Fix flaky ramp-up test.stefan@webrtc.org
Don't require the first estimate to be less than the target bitrate. There are other tests verifying that BWE works, so it's enough for this test to measure the time it takes to ramp-up. R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20979004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6764 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-20Check before send/receive rtp header extensions.pbos@webrtc.org
BUG=1788 R=pbos@webrtc.org, tommi@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13949004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6744 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15Make RTCP sender report send media bytes.pbos@webrtc.org
r6654 changed RtpSender::Bytes() to return the number of bytes sent instead of number of media bytes. This is used by VideoEngine for stats. This change broke RTCP which sends this same count as the number of payload bytes sent (excluding headers and padding). BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14959004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6691 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.stefan@webrtc.org
This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems. R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21869005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11Remove the send-side cname getter APIs from voice and video engine.stefan@webrtc.org
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname. R=henrika@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
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