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1fd9595936
Pass VideoDecoderParams to VideoDecoderFactory and add SSRC to RtpEncodingParameters.
2016-06-22 00:46:15 -07:00
a36350d790
Roll chromium_revision 0cb69fcbdc..118f252384 (401034:401159)
2016-06-21 19:39:53 -07:00
6ef36e708f
Remove DesktopCapturer::Callback::OnCaptureCompleted()
2016-06-21 16:50:00 -07:00
dfc4244d9e
Update ICE role on all ports, not just ones used for new connections.
2016-06-21 14:19:48 -07:00
123f33cd00
Revert of Delete method cricket::VideoFrame::Copy. (patchset #7 id:120001 of https://codereview.webrtc.org/2080253002/ )
2016-06-21 14:02:52 -07:00
0af180b1ae
When a remote candidate is added, update all prflx candidates.
2016-06-21 13:15:32 -07:00
7e4e00d189
Delete method cricket::VideoFrame::Copy.
2016-06-21 12:53:48 -07:00
3ee3d69e7d
Roll chromium_revision 719d4716ed..0cb69fcbdc (400938:401034)
2016-06-21 12:22:24 -07:00
7bd5f253bc
Fine tune the IntelligibilityEnhancer
2016-06-21 11:30:25 -07:00
435264a183
Increase the stun ping interval.
2016-06-21 11:28:38 -07:00
8d2248ff30
Revert of Update ICE role on all ports, not just ones used for new connections. (patchset #3 id:40001 of https://codereview.webrtc.org/2053043003/ )
2016-06-21 08:29:21 -07:00
03153f1032
GN: Add neteq_rtpplay and rtc_event_log_source
2016-06-21 05:38:42 -07:00
3a2a6404b1
Revert of Delete method cricket::VideoFrame::Copy. (patchset #7 id:120001 of https://codereview.webrtc.org/2080253002/ )
2016-06-21 04:17:28 -07:00
0ed2763403
Roll chromium_revision 12074d3d1a..719d4716ed (400873:400938)
2016-06-21 04:14:45 -07:00
9c00f646f0
Delete method cricket::VideoFrame::Copy.
2016-06-21 04:04:25 -07:00
1e6bbe4538
Delete deprecated VideoFrameBuffer methods.
2016-06-21 03:59:23 -07:00
41ed7e1715
Avoid race when stopping audio unit on iOS
2016-06-21 11:41:07 +02:00
86eff72eec
Adds logging in combination with restart of audio unit
2016-06-21 11:26:47 +02:00
c3a34ed544
Disable P2PTransportChannelTest.TestIceConfigWillPassDownToPort
2016-06-21 02:21:10 -07:00
69b34625c1
Exclude libjingle_peerconnection_{jni,so} targets from Chromium builds.
2016-06-21 01:05:16 -07:00
2e82f3821f
Reland of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #1 id:1 of https://codereview.webrtc.org/2084873002/ )
2016-06-21 00:26:43 -07:00
a536bfe70d
Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #5 id:340001 of https://codereview.webrtc.org/2078873002/ )
2016-06-21 00:08:49 -07:00
1eebc17f42
Roll chromium_revision 4a4b616f6a..12074d3d1a (400720:400873)
2016-06-20 19:39:38 -07:00
351da09467
Remove header files for the AEC and the APM test program that are no longer used.
2016-06-20 14:32:57 -07:00
370544594e
Update ICE role on all ports, not just ones used for new connections.
2016-06-20 12:55:52 -07:00
ce89cbd04a
Roll chromium_revision 6a85b3b953..4a4b616f6a (400641:400720)
2016-06-20 12:30:07 -07:00
d685fef94c
Use the new API to set the BoringSSL time callback.
2016-06-20 12:00:44 -07:00
2169d8bc68
Reland of move audio/video distinction for probe packets. (patchset #1 id:1 of https://codereview.webrtc.org/2086633002/ )
2016-06-20 11:53:02 -07:00
6d3e0c22c3
Use QualityScaler for OpenH264 encoder.
2016-06-20 20:49:32 +02:00
17bde8c96e
Revert of Remove audio/video distinction for probe packets. (patchset #2 id:20001 of https://codereview.webrtc.org/2061193002/ )
2016-06-20 11:47:16 -07:00
4b6c8b7bf7
Fix ProcessReverseStream usage in audioproc_f
2016-06-20 11:02:30 -07:00
884c336c34
Reland of IncomingVideoStream refactoring. This reland does not contain the non-smoothing part of the original implementation. Instead, when smoothing is turned off, frame callbacks run on the decoder thread, as they did before. This code path is used in Chrome. As far as Chrome goes, the difference now is that there won't be an instance of IncomingVideoStream in between the decoder and the callback (i.e. fewer locks). Other than that, no change for Chrome.
2016-06-20 19:42:44 +02:00
7ebbf90077
New rtc dump analyzing tool in Python
2016-06-20 07:39:15 -07:00
3e33bfeb6d
Fix some sign-compare warnings in webrtc/api.
2016-06-20 07:04:09 -07:00
839315beca
Use the Chromium libfuzzer template instead of rolling our own.
2016-06-20 06:03:39 -07:00
1a20610764
Fix buffer overflow in HMAC validation of STUN messages.
2016-06-20 05:13:16 -07:00
c853597598
rtc::Buffer: Grow capacity by at least 1.5x to prevent quadratic behavior
2016-06-20 04:47:39 -07:00
504f3351d2
Roll chromium_revision 465d55d04e..6a85b3b953 (400622:400641)
2016-06-20 03:55:08 -07:00
ac62bd4a3b
Rewrite CreateBlackFrame in webrtcvideoengine.
2016-06-20 03:38:52 -07:00
44bf02fba2
Remove SdpAudioFormat's default constructor
2016-06-20 02:39:48 -07:00
a7d88d3844
Remove audio/video distinction for probe packets.
2016-06-20 10:51:08 +02:00
02343b9ae2
Remove dead GYP target audio_device_module_java
2016-06-20 01:43:36 -07:00
442e6ee76a
Workaround java.gypi inclusion error in Chromium builds.
2016-06-20 01:34:06 -07:00
4c7f8aec41
Cleanup MIPS specific link configuration
2016-06-19 22:39:20 -07:00
fc36a2d558
Roll chromium_revision a21316a36e..465d55d04e (400620:400622)
2016-06-19 19:38:04 -07:00
ff702f5c4c
Roll chromium_revision f66fe7e469..a21316a36e (400617:400620)
2016-06-19 11:37:52 -07:00
71687f3c4d
Roll chromium_revision 5cdeb1b846..f66fe7e469 (400605:400617)
2016-06-19 03:39:49 -07:00
a9df50a3b3
Roll chromium_revision 0962148116..5cdeb1b846 (400593:400605)
2016-06-18 19:38:38 -07:00
ce5a874674
Improve encoding time calculation for Android HW encoder.
2016-06-18 19:12:57 -07:00
7508f3d7d4
Roll chromium_revision 6c3ee789f0..0962148116 (400588:400593)
2016-06-18 11:49:11 -07:00
5023d41cb4
GN: Update xmpp and p2p to match Chromium build
2016-06-18 04:23:01 -07:00
a935574b27
Roll chromium_revision e10a42e0d1..6c3ee789f0 (400570:400588)
2016-06-18 03:41:58 -07:00
cc5a5824f8
Roll chromium_revision b078b9902f..e10a42e0d1 (400452:400570)
2016-06-17 20:23:33 -07:00
61a6946a8c
Roll chromium_revision 3a1c71fcbd..b078b9902f (400409:400452)
2016-06-17 11:42:53 -07:00
f03a8d4c4d
Unpack different wav files after each INIT event of the aecdump
2016-06-17 09:41:41 -07:00
863a8264cc
Use |probe_cluster_id| to cluster packets.
2016-06-17 09:21:34 -07:00
217fb66e16
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
2016-06-17 08:30:54 -07:00
387000114d
Remove some dead code from VCMJitterBuffer.
2016-06-17 17:07:17 +02:00
57c21f9b44
Remove ViEEncoder::Pause / Start
2016-06-17 07:27:16 -07:00
c13ded54ca
Move AudioCodingModuleImpl to anonymous namespace in audio_coding_module.cc
2016-06-17 06:00:45 -07:00
434b85d708
Roll chromium_revision ed2c9cb4cb..3a1c71fcbd (400384:400409)
2016-06-17 05:31:37 -07:00
ca6d5d1c9f
Partial reland of Delete unused and almost unused frame-related methods. (patchset #1 id:1 of https://codereview.webrtc.org/2076113002/ )
2016-06-17 05:03:04 -07:00
fd634c43e9
Reland of Re-enable UBsan on AGC.
2016-06-17 04:36:10 -07:00
07ec26d1a9
Fix crash parsing malformed rtp packet where header extesnsion size mismatch expected.
2016-06-17 04:18:54 -07:00
9b99499124
Added a builtin audio decoder factory to the default PeerConnectionFactory constructor.
2016-06-17 04:16:24 -07:00
62379c89d0
Move Camera1 specific methods to Camera1Enumerator and create CameraEnumerator interface.
2016-06-17 03:45:45 -07:00
72e735d386
Revert of Delete unused and almost unused frame-related methods. (patchset #12 id:220001 of https://codereview.webrtc.org/2065733003/ )
2016-06-17 02:55:14 -07:00
76270de4bc
Delete unused and almost unused frame-related methods.
2016-06-17 02:00:12 -07:00
0dbc8bfbe5
Roll chromium_revision 1a73d11e65..ed2c9cb4cb (399420:400384)
2016-06-17 01:33:47 -07:00
e6c9e88c18
Android: Add Size class.
2016-06-17 01:02:04 -07:00
50c4821110
Fix missing resource file in webrtc_perf_tests.isolate
2016-06-17 00:39:41 -07:00
6af2e86b46
Refactor VideoDenoiser to work with I420Buffer, not VideoFrame.
2016-06-17 09:12:44 +02:00
68208897be
Roll chromium_revision 7fa6701bc5..1a73d11e65 (398458:399420)
2016-06-16 23:29:30 -07:00
d1523ca38f
Fix header size check in PseudoTcp::parse().
2016-06-16 18:08:12 -07:00
8e8222d0d2
Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #4 id:290001 of https://codereview.webrtc.org/2071473002/ )
2016-06-16 15:44:04 -07:00
e03f878737
Reland of Split IncomingVideoStream into two implementations, with smoothing and without.
2016-06-16 13:29:02 -07:00
4a0f7b508d
- Remove use of VoERTP_RTCP::SetLocalSSRC() for receive streams; recreate them instead. - Remove VoERTP_RTCP from VoEWrapper and FakeWebRtcVoiceEngine.
2016-06-16 13:07:33 -07:00
3abb764400
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 12:08:03 -07:00
9421853e17
Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().
2016-06-16 10:53:22 -07:00
e565a04de3
Revert of Fix crash parsing malformed rtp packet (patchset #1 id:1 of https://codereview.webrtc.org/2067793003/ )
2016-06-16 10:04:50 -07:00
5a45fe6fd7
Fix crash parsing malformed rtp packet where header extesnsion size mismatch expected.
2016-06-16 18:52:37 +02:00
4867ca2689
Revert of -enable UBsan on AGC. (patchset #1 id:1 of https://codereview.webrtc.org/2063643003/ )
2016-06-16 07:59:09 -07:00
30a3a751a6
Fix buffer overflow parsing malformed rtp packet that has one-byte length extension going past extensions block
2016-06-16 15:57:15 +02:00
1642620a7d
Performance fix for H264 RBSP parsing.
2016-06-16 15:54:44 +02:00
fc3a8ee47b
Delete unused code.
2016-06-16 15:51:31 +02:00
2d014be554
Resolves issue with bad audio using BT headsets on iOS.
2016-06-16 14:26:55 +02:00
5a9e7e0ba0
Fix a few error prone lines on VideoCapturerAndroid.
2016-06-16 14:25:55 +02:00
b00dc386d3
Delete GetExecutablePath and related unused code.
2016-06-16 12:44:30 +02:00
342f74005f
NetEq: Ask AudioDecoder for sample rate instead of passing it as an argument
2016-06-16 03:18:00 -07:00
347d35129e
AudioDecoder: Remove the default implementation of SampleRateHz
2016-06-16 01:59:09 -07:00
371b43b86a
Changes synchronization offset perfomance tracking to be logged after converge only.
2016-06-16 09:58:44 +02:00
4f0dfbd213
Change initial DTLS retransmission timer from 1 second to 50ms.
2016-06-15 17:15:23 -07:00
947c02d444
Disable WebRtcVideoChannel2BaseTest.AddRemoveCapturer because it is flaky
2016-06-15 15:39:46 -07:00
4c17abe35e
Add DesktopCapturer::Result::MAX_VALUE
2016-06-15 13:55:44 -07:00
14461d42bc
Fixing flaky test: WebRtcSessionTest.TestPacketOptionsAndOnPacketSent
2016-06-15 11:06:57 -07:00
a6219cc3ef
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 10:30:14 -07:00
74290b9d9c
New rtc dump analyzing tool in Python CQ_INCLUDE_TRYBOTS=tryserver.chromium.linux:linux_chromium_gn_upload R=henrik.lundin@webrtc.org , ivoc@webrtc.org , kwiberg@webrtc.org , peah@webrtc.org , phoglund@webrtc.org
2016-06-15 17:19:06 +02:00
ceb9d0cc29
Audio decoder factory test: Ensure that g722's sample rate is 16 kHz, not 8 kHz
2016-06-15 05:55:17 -07:00
6808419068
iSAC decoder: Remove obsolete TODO
2016-06-15 05:54:10 -07:00
edaa849013
WebRtcVoiceCodecs: Eliminate some useless copying
2016-06-15 04:34:47 -07:00
sakal