fix iOS build issue #164 #27

taste1981 · 2019-08-21T16:01:47Z

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The following overrides were missing: * OnPacketLossRateUpdate * OnRttUpdate * OnLossNotification Bug: webrtc:10501 Change-Id: I9b02d9cc153f2ad4cbf3c50ee3a17f3fa152da93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135561 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27873}

It's more convenient to have it next to other freeze-related metrics when sorted alphabetically. Bug: None Change-Id: I8f73e32a9cdb8166e139c193ffcd1dcc1fa18533 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135563 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27874}

Bug: None Change-Id: Iaf91218e3ebedf301e991083fe32cb26ba5b7476 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135562 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27875}

TBR=nisse@webrtc.org Bug: None Change-Id: I6cc0651a4d01e1d46941a6bb7ee97fdc98b11514 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135564 Reviewed-by: Elad Alon <eladalon@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27876}

This CL removes the usage of absl::flat_hash_map because it transitively depends on CCTZ which fails to link with lld-link after the switch to libc++. Since std::map doesn't support heterogeneous lookup until C++14, this CL also stops using absl::string_view and switches to `const std::string&`. Bug: webrtc:10605 Change-Id: I4fc93969c6fc0cc7e7e62b4d2f801bdd27cff0f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135566 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27877}

Preparation for deleting PeerConnectionInterface::StartRtcEventLog method with a PlatformFile argument. Bug: webrtc:6463 Change-Id: Ia9fa1d99a3d87f3bf193e73382690b782ffea65c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135285 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27879}

These metrics were previously collected by WebRTC, but not printed. Bug: None Change-Id: I79cf4b70da7608d88f13f21c92170d45d00ccaa5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135567 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27880}

This CL makes it possible to configure the priority of audio streams in bitrate allocations using field trials. It also adds the option to forcibly ignore any injected audio allocation strategy, so that experimentation with allocation won't be blocked on the work to remove the strategy injection. Bug: webrtc:10603 Change-Id: Ic36ceee6c15eb0fad275866f77e2a121066e516c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135467 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27881}

The state this set tracks (whether this is new feedback for a packet belonging to a media ssrc) can already be inferred from data provided by the SendTimeHistory: if packet send time is not populated in the feedback it's either because: 1. The feedback has already been processed 2. The receiver is sending feedback for bogus non-existent packets If the first case, this maps to |feedback_packet_seq_num_set_| containing the packet, if the ssrc (present in the feedback) is part of the media ssrcs. In the second case, this data should be ignored anyway. Bug: webrtc:10604 Change-Id: If4828091142d68baa8dbb62be9d0b24ccaaa9546 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135163 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27882}

These issues have been fixed upstream in Abseil. Bug: webrtc:10138 Change-Id: Ic0ebd22d0ad95bbd5269c08c182a76f9bf42f3a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135571 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27883}

The main purpose right now of this CL is to avoid the situation where multiple retransmissions are queued for sending (normally after network glitch with increased pacer queue length), and some of those fail sending because the can't be retrieved from the packet history due to too short time since last sent. Bug: webrtc:8975, webrtc:10607 Change-Id: I9f6369d83f0b8208e5f57b2dc2fd3f2db7c6fea1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135164 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27884}

Bug: none Change-Id: Ia662068fe179faebc1df0aaa7f37b6e989b6525f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135569 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27888}

Tested with a Web Platform Test; the test added here is useful, but does not exercise the bug. Bug: chromium:959128 Change-Id: Ia2e7f9e015b2345dd02d341b0fe27f58b64aa81e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135575 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27889}

Related code was deleted in https://webrtc-review.googlesource.com/c/112081. Bug: webrtc:10080 Change-Id: I3adc1238df6e80380cae3403c108403a59fd4a05 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135740 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27890}

Bug: webrtc:10608 Change-Id: Iac1111b739458a1b0ce1cac5e59de06905c085d2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135574 Commit-Queue: Fredrik Hernqvist <fhernqvist@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27891}

Bug: none Change-Id: Iad6dfdfdae13149bb8abe4b884e288e50aa7b73d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135102 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27892}

Moved to modules/remote_bitrate_estimator/overuse_estimator.h. Bug: webrtc:5876 Change-Id: Iae4b07d94bf4f16b887c3a4129168c4a45a3b5e1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135570 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27893}

…frames in one place"" This reverts commit 4fb12b0. Reason for revert: Breaks some asan chromium bots Original change's description: > Reland "Copy video frames metadata between encoded and plain frames in one place" > > Reland with fixes. > > Currently some video frames metadata like rotation or ntp timestamps are > copied in every encoder and decoder separately. This CL makes copying to > happen at a single place for send or receive side. This will make it > easier to add new metadata in the future. > > Also, added some missing tests. > > Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346 > > Bug: webrtc:10460 > Change-Id: Ia71198685de7fbd990704b575231cdce94dc0645 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134961 > Reviewed-by: Johannes Kron <kron@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27828} TBR=ilnik@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,kron@webrtc.org,artit@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:10460 Change-Id: I9c87a43a716622b389974cb8377f973573fc29a5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135747 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27895}

This is a reland of 14b2758 Original change's description: > Version 2 "Refactoring DataContentDescription class" > > (substantial changes since version 1) > > This CL splits the cricket::DataContentDescription class into > two classes: cricket::RtpDataContentDescription (used for RTP data) > and cricket::SctpDataContentDescription (used for SCTP only). > > SctpDataContentDescription no longer inherits from > MediaContentDescriptionImpl, and no longer contains "codecs". > > Due to usage of internal interfaces by consumers, shimming the old > DataContentDescription API is needed. > > A new cricket::DataContentDescription class is defined, which is > a shim over RtpDataContentDescription and SctpDataContentDescription. > It exposes as little functionality as possible, but supports the > concerned consumer's usage > > Design document: > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > Bug: webrtc:10358 > Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900 > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27853} Bug: webrtc:10358 Change-Id: Iff45c4694167f0b31b34ff2167c1f4ffa650bcc4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135281 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27896}

Methods of P2PTransportChannel have been assuming a non-null port allocator for a long time, and yet the constructor does not check for that. With the recent change that wires a signal in the port allocator to the transport in the constructor, a valid allocator becomes a must. Bug: None Change-Id: I4ec2e5b577d74a598ee3c2f8ad59e9f0285ac4b9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135880 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Qingsi Wang <qingsi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27897}

This reverts commit bd20c3f. Reason for revert: chromium:961253 This CL is not the cause of the regression, but reverting it will make the reverting of the actual cause easier. Original change's description: > Rename configurations_ to vpx_configs_ in LibvpxVp8Encoder > > Bug: None > Change-Id: I548a724f0fb81f46785517c90e527edc075e1476 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135040 > Commit-Queue: Elad Alon <eladalon@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27836} TBR=brandtr@webrtc.org,eladalon@webrtc.org Bug: chromium:961253 Change-Id: I707337e0ce50f29f9cda7cf45500c11debace1a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135750 Reviewed-by: Elad Alon <eladalon@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27898}

…ferController" This reverts commit 4d6795f. Reason for revert: chromium:961253 Original change's description: > Refactor handling of configuration overrides from Vp8FrameBufferController > > Make Vp8FrameBufferController::UpdateConfiguration return a set > of desired overrides. These overrides are cumulative with > previously returned override sets. > > Bug: webrtc:10382 > Change-Id: I1aa9544ae0cf6c57115e80963b3bbcdc3101db5e > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134649 > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Commit-Queue: Elad Alon <eladalon@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27835} TBR=brandtr@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org Bug: chromium:961253 Change-Id: I06f0eafd4f38c441ddbdfeebae8055b02465eb9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135940 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27900}

Bug: webrtc:6424 Change-Id: I966715f3d815d85b3ec46333cea80418944cd4b1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135742 Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27902}

Bug: webrtc:10450 Change-Id: I6f586a3c6781450b9bfdcc31dc3f49f6289d70e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138265 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28096}

If screen share is set, then we need to tell video source, that it is screen share source. Also video track should be aware, that it is screen share track. It is required to choose proper video encoding settings. Bug: webrtc:10138 Change-Id: I5c82584ae0325a303a495554d87962a98b676694 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138278 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28098}

Plus tests for 16 kHz. Bug: webrtc:10631 Change-Id: I2d89bc6d0d9548f0ad7bb1e36d6dfde6b6b31f83 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138072 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28099}

Bug: webrtc:10559 Change-Id: I956287e71a47856cfb6dd807d9715d6ee2572f55 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138263 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28100}

This patch is a NOP and moves - class Connection - class ConnectionInfo - class ProxyConnection from port.{h/cc} to a new file called connection.{h/cc} BUG=webrtc:10647 Change-Id: I89322d3421d272657e24a46b28ab6679fcdc9450 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137509 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28101}

This is a partial revert of https://webrtc-review.googlesource.com/c/src/+/130101. The KeyFrameRequestSender argument is added back to the constructor of RtpVideoStreamReceiver. It is optional; if a null pointer is passed, key frame requests are sent via the internal RtpRtcp module, and this is how the class is used by VideoReceiveStream. An injectable KeyFrameRequestSender is useful for tests, for downstream applications that want to route key frame requests elsewhere, and should also aid later migration to RtcpTransciever. Bug: None Change-Id: Idf9baeed21570625ad74e9afbe38f7ea5bf79feb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139107 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28102}

Bug: webrtc:10206 Change-Id: Iabf3eea20027ae42faadabba572dc68f9b678121 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139105 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28103}

In addition to the 48 kHz that we've always used. Bug: webrtc:10631 Change-Id: If73bf7ff9c1c0d22e0d1caa245128612850f8e41 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138268 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28104}

Change-Id: Ie01d66d459f704e7fa99b439dd6f917e4e41cead Bug: webrtc:10698 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139106 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28105}

Bug: webrtc:10676 Change-Id: Ifdef133e123a0c54204397fb323f4c671c40a464 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135881 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28106}

…ests Bug: webrtc:10138, webrtc:10692 Change-Id: I4d7ae84dc2945fef6451a6671786b3b19cd9abd8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139108 Commit-Queue: Artem Titov <titovartem@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28107}

Recent kernel commit[1] moved a bit the define for this constant. This revealed a missing include in WebRTC. [1]: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?id=0768e17073dc5 Bug: webrtc:10677 Change-Id: I6ed69d307599d077760ae6ad74be10bfbdd1cac6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138270 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28108}

Bug: webrtc:9719 Change-Id: Ib394e02a4f0dcb36da64b9f005f068a53a50854c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139280 Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28111}

It does not make sense for DtlsTransport to own ICE, and this arrangement will not work when negotiating datagram or DTLS transport. During negotiation, both a DTLS transport and a datagram transport need to be ready to receive from the same ICE transport, depending on which protocol is chosen by the answerer. Once the answerer chooses a protocol, the transport that is not chosen must be deleted, but ICE must be left intact for use by the remaining transport. Bug: webrtc:9719 Change-Id: Ibab969b574c981e3834ced71f8ff88008cb26a6c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139340 Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28113}

Whenever a datagram is acked, the datagram transport will provide the remote peer's receive timestamp in this field. Bug: webrtc:9719 Change-Id: I516b9d602e62179a3deda001e0ee9b484aa20d37 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139440 Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28114}

Per discussions at https://crbug.com/webrtc/10753, the remote-outbound-rtp.ssrc is supposed to reflect the SSRC of the RTP media stream (i.e. outbound-rtp.ssrc) and not the sender that the corresponding RTCP report block was transmitted on. (cherry picked from commit 8605fbf) Bug: webrtc:10753 Change-Id: Id88f5fdbe6397ba81a46f0ef430bd6f08e66b145 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143484 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#28354} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143793 Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/branch-heads/m76@{#1} Cr-Branched-From: 57dc02a-refs/heads/master@{#28114}

…nd decoders This partially reverts this CLs: Reland "Copy video frames metadata between encoded and plain frames in one place" https://webrtc.googlesource.com/src/+/2ebf5239782bf6b46d4aa812f34fa9f9e5a02be9 The problem with this Cl is that ColorSpace was made to always be copied from the EncodedImage in the GenericDecoder, which overwrote ColorSpace information from the decoder. If decoder applied color space transition or bitstream color space information was different from the WebRTC signaled one, the incorrect color space data were passed to the renderer. This CL removes introduced change regarding color space data: GenericDecoder doesn't copy or store it and software decoders are restored to copy it. Relevant tests are also removed. cherry-picked from https://webrtc.googlesource.com/src/+/a0e2609a08d0771d4dde6d5a94b06b71f0cdf53b Bug: chromium:982486 Change-Id: I989e01476ff7f7df376c05578ab8f540b95a1dd2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145331 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/branch-heads/m76@{open-webrtc-toolkit#2} Cr-Branched-From: 57dc02a-refs/heads/master@{#28114}

…nes if the default connection address is a hostname candidate. Using a FQDN in the c= line has caused an inter-op issue with Firefox when hostname candidates are the only candidates gathered when forming the media sections. To address this issue, we use 0.0.0.0:9 when a hostname candidate would be used to populate the c= and the m= lines. The SDP grammar related to ICE candidates has been moved out of RFC8445, and is currently defined in draft-ietf-mmusic-ice-sip-sdp. A FQDN address must not be used in the connection address attribute per the latest draft, if the ICE agent generates local candidates. Also, the wildcard addresses (0.0.0.0 or ::) with port 9 are given the exception as the connection address that will not result in an ICE mismatch. We thus adopt the aforementioned solution after combining these considerations. TBR=hta@webrtc.org (cherry picked from commit 3ae59d3) Bug: chromium:927309, chromium:982108 Change-Id: I3df2db0f154276da39f99650289cf81baa677e74 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145280 Commit-Queue: Qingsi Wang <qingsi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#28547} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145703 Reviewed-by: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/branch-heads/m76@{open-webrtc-toolkit#3} Cr-Branched-From: 57dc02a-refs/heads/master@{#28114}

Change-Id: Ic1eb84eaa0e59c9c768bb5575946c9f47c2bd224 Reviewed-on: https://git-ccr-1.devtools.intel.com/gerrit/37117 Reviewed-by: webrtctest <webrtctest@intel.com> Tested-by: webrtctest <webrtctest@intel.com> Reviewed-by: Xian, Yuqiang <yuqiang.xian@intel.com>

This reverts commit c5f7108. Reason for revert: Causes Chromium WPT Tests to fail, preventing rolls. Sample failed run: https://ci.chromium.org/p/chromium/builders/try/linux-rel/511995? Sample logs: STDERR: # Fatal error in: ../../third_party/webrtc/pc/peer_connection.cc, line 575 STDERR: # last system error: 0 STDERR: # Check failed: (signaling_thread())->IsCurrent() STDERR: # Received signal 6 STDERR: #0 0x7f81d39e3de9 base::debug::CollectStackTrace() STDERR: #1 0x7f81d38f9ca3 base::debug::StackTrace::StackTrace() STDERR: #2 0x7f81d39e393b base::debug::(anonymous namespace)::StackDumpSignalHandler() STDERR: #3 0x7f81c9054140 (/lib/x86_64-linux-gnu/libpthread-2.31.so+0x1413f) STDERR: #4 0x7f81c8d72db1 gsignal STDERR: #5 0x7f81c8d5c537 abort STDERR: #6 0x7f81c7344032 rtc::webrtc_checks_impl::FatalLog() STDERR: #7 0x7f81c722e5c0 webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>() STDERR: #8 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach() STDERR: #9 0x7f81c72d6e8e webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>() STDERR: #10 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach() STDERR: #11 0x7f81c71c6df3 webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>() STDERR: #12 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach() STDERR: #13 0x7f81c73135bc rtc::OpenSSLStreamAdapter::ContinueSSL() STDERR: #14 0x7f81c7312fd4 rtc::OpenSSLStreamAdapter::OnEvent() STDERR: #15 0x7f81c71c30d9 cricket::StreamInterfaceChannel::OnPacketReceived() STDERR: #16 0x7f81c71c640a cricket::DtlsTransport::OnReadPacket() STDERR: #17 0x7f81c71cad61 cricket::P2PTransportChannel::OnReadPacket() STDERR: #18 0x7f81c71bc90f cricket::Connection::OnReadPacket() STDERR: #19 0x7f81c71e6255 cricket::UDPPort::HandleIncomingPacket() STDERR: #20 0x7f81cd1f1bff blink::(anonymous namespace)::IpcPacketSocket::OnDataReceived() STDERR: #21 0x7f81cd1f645d blink::P2PSocketClientImpl::DataReceived() STDERR: #22 0x7f81cd50a16b network::mojom::blink::P2PSocketClientStubDispatch::Accept() STDERR: #23 0x7f81d2b4f642 mojo::InterfaceEndpointClient::HandleValidatedMessage() STDERR: #24 0x7f81d2b5304b mojo::MessageDispatcher::Accept() STDERR: #25 0x7f81d2b50bb1 mojo::InterfaceEndpointClient::HandleIncomingMessage() STDERR: #26 0x7f81d2b58a3a mojo::internal::MultiplexRouter::ProcessIncomingMessage() STDERR: #27 0x7f81d2b57f7f mojo::internal::MultiplexRouter::Accept() STDERR: #28 0x7f81d2b5304b mojo::MessageDispatcher::Accept() STDERR: #29 0x7f81d2b48851 mojo::Connector::DispatchMessage() STDERR: #30 0x7f81d2b492e7 mojo::Connector::ReadAllAvailableMessages() STDERR: #31 0x7f81d2b490a3 mojo::Connector::OnHandleReadyInternal() STDERR: #32 0x7f81d2b498f0 mojo::SimpleWatcher::DiscardReadyState() STDERR: #33 0x7f81d2d0e67a mojo::SimpleWatcher::OnHandleReady() STDERR: #34 0x7f81d2d0eb2b base::internal::Invoker<>::RunOnce() STDERR: #35 0x7f81d397f85b base::TaskAnnotator::RunTask() STDERR: #36 0x7f81d399a04c base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWorkImpl() STDERR: #37 0x7f81d3999c78 base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWork() STDERR: #38 0x7f81d391fe64 base::MessagePumpDefault::Run() STDERR: #39 0x7f81d399a8dc base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::Run() STDERR: #40 0x7f81d395ae55 base::RunLoop::Run() STDERR: #41 0x7f81d39c87f2 base::Thread::Run() Original change's description: > Reland "Replace sigslot usages with robocaller library." > > This is a reland of 40261c3 > > Note: Instead of changing the type of JsepTransportController->SignalSSLHandshakeError > added a new member with a different name and used it in webrtc code. > After this change do two more follow up CLs to completely remove the old code > from google3. > > Original change's description: > > Replace sigslot usages with robocaller library. > > > > - Replace all the top level signals from jsep_transport_controller. > > - There are still sigslot usages in this file so keep the inheritance > > and that is the reason for not having a binary size gain in this CL. > > > > Bug: webrtc:11943 > > Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540 > > Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#32321} > > Bug: webrtc:11943 > Change-Id: Ia07394ee395f94836f6b576c3a97d119a7678e1a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186946 > Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32359} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,glahiru@webrtc.org Change-Id: I6bce1775d10758ac4a9d05b855f473dd70bd9815 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:11943 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187487 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32372}

Elad Alon and others added 30 commits May 8, 2019 08:01

chromium-webrtc-autoroll and others added 28 commits May 29, 2019 02:30

Some changes for building with OWT. (open-webrtc-toolkit#13)

9ca8f80

Update webrtc visiblity to SDK (open-webrtc-toolkit#23)

ae00694

Rebase all changes for HEVC to M70

6c5fb02

Fix ios build issue open-webrtc-toolkit#164

a7f6e53

taste1981 closed this Aug 21, 2019

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fix iOS build issue #164 #27

fix iOS build issue #164 #27

taste1981 commented Aug 21, 2019

fix iOS build issue #164 #27

fix iOS build issue #164 #27

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taste1981 commented Aug 21, 2019