961168a6dbAdd sakal as an OWNER to some Android files.
sakal
2016-09-09 00:22:25 -07:00
ce2e13602eUpdate AvgCounter to have the ability to include last period metric for subsequent intervals without samples (e.g. for non-periodic updated stats).
asapersson
2016-09-09 00:13:35 -07:00
2a5f371df3Make UMA stats creation available in the Java interface.
sakal
2016-09-09 00:11:48 -07:00
9365338db2Only expose gflags target in non-Chromium and non-fuzzer builds.
kjellander
2016-09-09 00:05:58 -07:00
aa85cac7b4Add magjed@ as owner of webrtc/common_video
magjed
2016-09-08 14:15:29 -07:00
906f403088This CL refactors the buffering of the incoming near-end signal inside the AEC. This solves the following issues: -Even though the buffering was previously done using ringbuffers, those were inefficiently used which caused a lot of hidden memcopys. -The ringbuffers wasted a lot of space in the AEC state as they were too long. -The lowest and two upper bands were decoupled in the buffering, which required extra code to handle. -On top of the ringbuffers there was a second linear buffer that was stored in the state which caused even more data to be stored on the state. -The incoming nearend frames were passed to the functions in the form of buffers on the state, which made the code harder to read as it was not immediately clear where the nearend signal was used, and when it was modified.
peah
2016-09-08 09:49:42 -07:00
0e62f2bebeChange owner of webrtc/test/channel_transport to solenberg@.
henrikg
2016-09-08 08:41:50 -07:00
f07fb00131Separating video settings in VideoQualityTest.
minyue
2016-09-08 08:20:07 -07:00
3115b06957Add a DEPS gclient hook to prune corrupt mockito remote.
ehmaldonado
2016-09-08 08:02:08 -07:00
a4c2106120This CL contains the following small changes:
aleloi
2016-09-08 01:25:46 -07:00
250fd97a67Use RateCounter for received bitrate stats: "WebRTC.Call.BitrateReceivedInKbps" "WebRTC.Call.VideoBitrateReceivedInKbps" "WebRTC.Call.AudioBitrateReceivedInKbps" "WebRTC.Call.RtcpBitrateReceivedInBps"
asapersson
2016-09-08 00:07:21 -07:00
14f1250f47Do not report bucket delay for stats when pacer is paused (zero returned).
asapersson
2016-09-07 23:14:50 -07:00
a264ecc456Copy RTCAudioSource.h and RTCMediaSource.h with other public header files when building the WebRTC framework for iOS / Mac
VladimirTechMan
2016-09-07 23:11:25 -07:00
14b9d79bd6If encoding is inactive, don't start sending when stream is reconfigured.
Taylor Brandstetter
2016-09-07 17:16:54 -07:00
4016a0b2e8GN: Move variables from //build_overrides/webrtc.gni to //webrtc/build/webrtc.gni
ehmaldonado
2016-09-07 06:50:19 -07:00
d52063fb07Change OverUseFrameDetector to use a task queue instead of ProcessThread to periodically check for overuse. It is made to only operate on a single task queue.
perkj
2016-09-07 06:32:18 -07:00
311525e715Several lock acquisitions and one of the two lock members are removed. ENSURE_LOCKS_REQUIRED and CalledOnValidThread annotations are added.
aleloi
2016-09-07 06:13:12 -07:00
08b0351dddImplement PlayoutDelay extension as a trait to be used with rtp::Packet class
Danil Chapovalov
2016-09-07 15:08:13 +02:00
10e8f8e2a4Relax expectation in EndToEndTest.CallReportsRttForSender test to reduce flakiness by ignoring potentional rounding errors and minor ntp time adjustments.
Danil Chapovalov
2016-09-07 15:05:29 +02:00
2d273f1e97Setting up an RTP input fuzzer for NetEq
henrik.lundin
2016-09-07 05:57:27 -07:00
947c66ccbbCQ: Remove linux_baremetal until it's back
Henrik Kjellander
2016-09-07 14:51:52 +02:00
9881cb2874Merge min_ms and max_ms accessors in PlayoutDelayOracle to reduce CriticalSection enterencies and avoid potentional synchronisation issues.
Danil Chapovalov
2016-09-07 13:29:47 +02:00
126ee727a0Only parse PPS up to PPS and SPS ids in the depacketizater.
Stefan Holmer
2016-09-07 12:46:21 +02:00
d4626e5f1eDisable -Wsentinel warning for Linux 32-bit builds.
kjellander
2016-09-07 02:33:01 -07:00
60e4346955Add time line for acked bitrate.
Stefan Holmer
2016-09-07 09:58:20 +02:00
b9d8d10d42Fixed flaky StunRequestTests which depended on the wall clock
skvlad
2016-09-06 17:17:17 -07:00
b460fd8b16Increase timeout for flaky tests for ProcessThreadImpl
skvlad
2016-09-06 14:34:32 -07:00
cfaca03b4bAdd dynamic bitrate tracker and adjustment for Exynos VP8 HW encoder.
Alex Glaznev
2016-09-06 14:08:19 -07:00
37062ed16fFix chromium-style errors in IntelligibilityEnhancer
Alejandro Luebs
2016-09-06 11:25:40 -07:00
857a8fb861Remove dedicated unittest file for remb format RembStatus moved to RtcpSender unittest where it fits better Creating remb in Compound/ReducedSize modes already covered by RtcpSender unittests. Parsing remb already covered by RtcpReceiverTest.ReceivesRemb
Danil Chapovalov
2016-09-06 11:41:46 +02:00
dedc822830MB: Cleanup no-longer-used GN configurations
kjellander
2016-09-06 02:32:33 -07:00
edaa4a45efGN: Set WEBRTC_RESTRICT_LOGGING as is set in GYP.
ehmaldonado
2016-09-06 02:27:47 -07:00
859e86107dRemove stop method from VideoTrackSourceInterface.
sakal
2016-09-06 02:25:08 -07:00
c928453fd0Significantly increased max_num_buffers_ of Vp9FrameBufferPool.
hbos
2016-09-06 01:38:28 -07:00
d4ec970a79neteq_rtpplay: Add an error message for unmatched SSRC
henrik.lundin
2016-09-06 01:22:45 -07:00
891eaac4fcMB: Add Linux 32-bit Debug and Release trybots
Henrik Kjellander
2016-09-06 09:52:45 +02:00
dbd8b6bec4Remove all reference to carbon api
kthelgason
2016-09-06 00:38:18 -07:00
0b1b472c51Enable gyp_compare_flag_script to compare all targets.
ehmaldonado
2016-09-05 23:19:46 -07:00
b27c0918f2Exclude flaky PhysicalSocketTest tests on Dr Memory bots.
Henrik Kjellander
2016-09-06 07:38:30 +02:00
aa90b313b4[WebRTC] Two DirectX capturers cannot work concurrently
zijiehe
2016-09-05 15:40:00 -07:00
fef8653c5aAn early analysis shows in DirectX based capturer, Windows API returns larger dirty region than the real screen change. A similar behavior may happen on other platforms with damage notification support. So it's better to have an individual layer to handle the Differ logic, and remove capturing independent logic out of each ScreenCapturer* implementation.
zijiehe
2016-09-05 15:26:32 -07:00
f71d298217GN: Apply optimize_max only on windows
ehmaldonado
2016-09-05 10:03:57 -07:00
1e714ae7a6RtcpReceiverTest rewritten using public available interface IncomingPacket(const uint8_t*, size_t) is used as entry point instead of IncomingRTCPPacket(PacketInformation* out, RtcpParser* in); Result is validated by checking which callbacks were called instead of checking intermediate structure PacketInformaion.
danilchap
2016-09-05 09:57:22 -07:00
d313403564rtc::Buffer: Let SetData and AppendData accept anything with .data() and .size()
kwiberg
2016-09-05 07:46:20 -07:00
84c8528f1eUse RtpPacketToSend in RtpSenderAudio. this eliminates reparsing of rtp packet on send audio path
danilchap
2016-09-05 07:27:57 -07:00
e9cc686293GN Templates: Move common_inherited_config to the template.
ehmaldonado
2016-09-05 06:10:18 -07:00
bd4317263aRestrict the 1-argument ArrayView constructor to types with .size() and .data()
kwiberg
2016-09-05 04:20:54 -07:00
59af8b7714GN Templates: Use the optimize_max compiler config.
ehmaldonado
2016-09-05 02:48:54 -07:00
ccd7489661Plot accumelated packets over time.
philipel
2016-09-05 02:46:25 -07:00
7556282a6cFrameBuffer::NextFrame now return a ReturnReason and take an additional std::unique_ptr<FrameObject>* as output parameter.
philipel
2016-09-05 10:57:41 +02:00
c82f2e1613RTCStatsCollector collecting stats on multiple threads.
hbos
2016-09-05 01:36:50 -07:00
7a2ce0b738GN Templates: Move common_config to the template.
ehmaldonado
2016-09-05 01:35:44 -07:00
74c10b5a7aIntroduce webrtc::VideoFrame::timestamp_us, and corresponding constructor. Replaces render_time_ms_, but old accessors are kept for compatibility.
nisse
2016-09-05 00:51:16 -07:00
3821ff8df5Only log BWE updates if the actual estimate changed or if we have non-zero loss reports.
stefan
2016-09-04 05:07:26 -07:00
a973f95386Remove restart method from VideoTrackSourceInterface.
sakal
2016-09-03 10:29:07 -07:00
9d1c54ace0[WebRTC] A real ScreenCapturer test
zijiehe
2016-09-02 19:10:34 -07:00
aecd982a6fChange a few configurations in AggressiveConfiguration Set bundle policy to max bundle. Set rtcp mux policy to required. Set enable ice renomination to true. This configuration is used by native applications.
Honghai Zhang
2016-09-02 16:58:17 -07:00
1afdf99f9cEventLogParser: use std::vector to reduce stack usage
skvlad
2016-09-02 13:22:26 -07:00
c55c39ddf7Limit initial fps value used in Android HW encoder initialization.
Alex Glaznev
2016-09-02 12:16:27 -07:00
79c6458575Provide a default implementation for PeerConnectionInterface::ice_state().
johan
2016-09-02 12:07:38 -07:00
84fcbd429fMB: Enable rtc_use_h264 for GYP builds for now.
kjellander
2016-09-02 11:55:39 -07:00
5e57b17283Introduce helpers to RtpSender to propagate RtpPacketToSend. The helpers intended to replace and deprecate BuildRtpHeader when RtpSenderAudio/RtpSenderVideo will be updated to pass RtpPacket class instead of raw buffer for sending.
Danil Chapovalov
2016-09-02 19:15:59 +02:00
hbos