webrtc_m130/pc/peerconnection_rtp_unittest.cc

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Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
/*
* Copyright 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <memory>
#include <vector>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/fakemetricsobserver.h"
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
#include "api/jsep.h"
#include "api/mediastreaminterface.h"
#include "api/peerconnectioninterface.h"
#include "pc/mediastream.h"
#include "pc/mediastreamtrack.h"
#include "pc/peerconnectionwrapper.h"
#include "pc/sdputils.h"
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
#include "pc/test/fakeaudiocapturemodule.h"
#include "pc/test/mockpeerconnectionobservers.h"
#include "rtc_base/checks.h"
#include "rtc_base/gunit.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/refcountedobject.h"
#include "rtc_base/scoped_ref_ptr.h"
#include "rtc_base/thread.h"
Reland "Add AddTransceiver and GetTransceivers to PeerConnection" This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80. Original change's description: > Revert "Add AddTransceiver and GetTransceivers to PeerConnection" > > This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a. > > Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout > > Original change's description: > > Add AddTransceiver and GetTransceivers to PeerConnection > > > > WebRTC 1.0 has added the transceiver API to PeerConnection. This > > is the first step towards exposing this to WebRTC consumers. For > > now, transceivers can be added and fetched but there is not yet > > support for creating offers/answers or setting local/remote > > descriptions. That support ("Unified Plan") will be added in > > follow-up CLs. > > > > The transceiver API is currently only available if the application > > opts in by specifying the kUnifiedPlan SDP semantics when creating > > the PeerConnection. > > > > Bug: webrtc:7600 > > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e > > Reviewed-on: https://webrtc-review.googlesource.com/23880 > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20896} > > TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org > > Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7600 > Reviewed-on: https://webrtc-review.googlesource.com/26400 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20897} TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7600 Reviewed-on: https://webrtc-review.googlesource.com/26401 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20898}
2017-11-27 13:01:52 -08:00
#include "test/gmock.h"
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
// This file contains tests for RTP Media API-related behavior of
// |webrtc::PeerConnection|, see https://w3c.github.io/webrtc-pc/#rtp-media-api.
Reland "Add AddTransceiver and GetTransceivers to PeerConnection" This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80. Original change's description: > Revert "Add AddTransceiver and GetTransceivers to PeerConnection" > > This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a. > > Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout > > Original change's description: > > Add AddTransceiver and GetTransceivers to PeerConnection > > > > WebRTC 1.0 has added the transceiver API to PeerConnection. This > > is the first step towards exposing this to WebRTC consumers. For > > now, transceivers can be added and fetched but there is not yet > > support for creating offers/answers or setting local/remote > > descriptions. That support ("Unified Plan") will be added in > > follow-up CLs. > > > > The transceiver API is currently only available if the application > > opts in by specifying the kUnifiedPlan SDP semantics when creating > > the PeerConnection. > > > > Bug: webrtc:7600 > > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e > > Reviewed-on: https://webrtc-review.googlesource.com/23880 > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20896} > > TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org > > Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7600 > Reviewed-on: https://webrtc-review.googlesource.com/26400 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20897} TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7600 Reviewed-on: https://webrtc-review.googlesource.com/26401 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20898}
2017-11-27 13:01:52 -08:00
namespace webrtc {
using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
using ::testing::ElementsAre;
using ::testing::UnorderedElementsAre;
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
const uint32_t kDefaultTimeout = 10000u;
template <typename MethodFunctor>
class OnSuccessObserver : public rtc::RefCountedObject<
webrtc::SetRemoteDescriptionObserverInterface> {
public:
explicit OnSuccessObserver(MethodFunctor on_success)
: on_success_(std::move(on_success)) {}
// webrtc::SetRemoteDescriptionObserverInterface implementation.
void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override {
RTC_CHECK(error.ok());
on_success_();
}
private:
MethodFunctor on_success_;
};
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
class PeerConnectionRtpTest : public testing::Test {
public:
PeerConnectionRtpTest()
Reland "Add AddTransceiver and GetTransceivers to PeerConnection" This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80. Original change's description: > Revert "Add AddTransceiver and GetTransceivers to PeerConnection" > > This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a. > > Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout > > Original change's description: > > Add AddTransceiver and GetTransceivers to PeerConnection > > > > WebRTC 1.0 has added the transceiver API to PeerConnection. This > > is the first step towards exposing this to WebRTC consumers. For > > now, transceivers can be added and fetched but there is not yet > > support for creating offers/answers or setting local/remote > > descriptions. That support ("Unified Plan") will be added in > > follow-up CLs. > > > > The transceiver API is currently only available if the application > > opts in by specifying the kUnifiedPlan SDP semantics when creating > > the PeerConnection. > > > > Bug: webrtc:7600 > > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e > > Reviewed-on: https://webrtc-review.googlesource.com/23880 > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20896} > > TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org > > Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7600 > Reviewed-on: https://webrtc-review.googlesource.com/26400 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20897} TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7600 Reviewed-on: https://webrtc-review.googlesource.com/26401 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20898}
2017-11-27 13:01:52 -08:00
: pc_factory_(
CreatePeerConnectionFactory(rtc::Thread::Current(),
rtc::Thread::Current(),
rtc::Thread::Current(),
FakeAudioCaptureModule::Create(),
CreateBuiltinAudioEncoderFactory(),
CreateBuiltinAudioDecoderFactory(),
nullptr,
nullptr)) {}
std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection() {
return CreatePeerConnection(RTCConfiguration());
}
std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWithPlanB() {
RTCConfiguration config;
config.sdp_semantics = SdpSemantics::kPlanB;
return CreatePeerConnection(config);
}
Reland "Add AddTransceiver and GetTransceivers to PeerConnection" This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80. Original change's description: > Revert "Add AddTransceiver and GetTransceivers to PeerConnection" > > This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a. > > Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout > > Original change's description: > > Add AddTransceiver and GetTransceivers to PeerConnection > > > > WebRTC 1.0 has added the transceiver API to PeerConnection. This > > is the first step towards exposing this to WebRTC consumers. For > > now, transceivers can be added and fetched but there is not yet > > support for creating offers/answers or setting local/remote > > descriptions. That support ("Unified Plan") will be added in > > follow-up CLs. > > > > The transceiver API is currently only available if the application > > opts in by specifying the kUnifiedPlan SDP semantics when creating > > the PeerConnection. > > > > Bug: webrtc:7600 > > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e > > Reviewed-on: https://webrtc-review.googlesource.com/23880 > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20896} > > TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org > > Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7600 > Reviewed-on: https://webrtc-review.googlesource.com/26400 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20897} TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7600 Reviewed-on: https://webrtc-review.googlesource.com/26401 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20898}
2017-11-27 13:01:52 -08:00
std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWithUnifiedPlan() {
RTCConfiguration config;
config.sdp_semantics = SdpSemantics::kUnifiedPlan;
return CreatePeerConnection(config);
}
std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection(
const RTCConfiguration& config) {
auto observer = rtc::MakeUnique<MockPeerConnectionObserver>();
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
auto pc = pc_factory_->CreatePeerConnection(config, nullptr, nullptr,
observer.get());
Reland "Add AddTransceiver and GetTransceivers to PeerConnection" This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80. Original change's description: > Revert "Add AddTransceiver and GetTransceivers to PeerConnection" > > This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a. > > Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout > > Original change's description: > > Add AddTransceiver and GetTransceivers to PeerConnection > > > > WebRTC 1.0 has added the transceiver API to PeerConnection. This > > is the first step towards exposing this to WebRTC consumers. For > > now, transceivers can be added and fetched but there is not yet > > support for creating offers/answers or setting local/remote > > descriptions. That support ("Unified Plan") will be added in > > follow-up CLs. > > > > The transceiver API is currently only available if the application > > opts in by specifying the kUnifiedPlan SDP semantics when creating > > the PeerConnection. > > > > Bug: webrtc:7600 > > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e > > Reviewed-on: https://webrtc-review.googlesource.com/23880 > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20896} > > TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org > > Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7600 > Reviewed-on: https://webrtc-review.googlesource.com/26400 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20897} TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7600 Reviewed-on: https://webrtc-review.googlesource.com/26401 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20898}
2017-11-27 13:01:52 -08:00
return rtc::MakeUnique<PeerConnectionWrapper>(pc_factory_, pc,
std::move(observer));
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
}
protected:
Reland "Add AddTransceiver and GetTransceivers to PeerConnection" This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80. Original change's description: > Revert "Add AddTransceiver and GetTransceivers to PeerConnection" > > This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a. > > Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout > > Original change's description: > > Add AddTransceiver and GetTransceivers to PeerConnection > > > > WebRTC 1.0 has added the transceiver API to PeerConnection. This > > is the first step towards exposing this to WebRTC consumers. For > > now, transceivers can be added and fetched but there is not yet > > support for creating offers/answers or setting local/remote > > descriptions. That support ("Unified Plan") will be added in > > follow-up CLs. > > > > The transceiver API is currently only available if the application > > opts in by specifying the kUnifiedPlan SDP semantics when creating > > the PeerConnection. > > > > Bug: webrtc:7600 > > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e > > Reviewed-on: https://webrtc-review.googlesource.com/23880 > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20896} > > TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org > > Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7600 > Reviewed-on: https://webrtc-review.googlesource.com/26400 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20897} TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7600 Reviewed-on: https://webrtc-review.googlesource.com/26401 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20898}
2017-11-27 13:01:52 -08:00
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
};
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
// These tests cover |webrtc::PeerConnectionObserver| callbacks firing upon
// setting the remote description.
class PeerConnectionRtpCallbacksTest : public PeerConnectionRtpTest {};
TEST_F(PeerConnectionRtpCallbacksTest, AddTrackWithoutStreamFiresOnAddTrack) {
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
ASSERT_TRUE(caller->AddTrack(caller->CreateAudioTrack("audio_track")));
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
// TODO(hbos): When "no stream" is handled correctly we would expect
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
// |add_track_events_[0].streams| to be empty. https://crbug.com/webrtc/7933
auto& add_track_event = callee->observer()->add_track_events_[0];
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_EQ(add_track_event.streams.size(), 1u);
EXPECT_TRUE(add_track_event.streams[0]->FindAudioTrack("audio_track"));
EXPECT_EQ(add_track_event.streams, add_track_event.receiver->streams());
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
}
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
TEST_F(PeerConnectionRtpCallbacksTest, AddTrackWithStreamFiresOnAddTrack) {
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
ASSERT_TRUE(caller->AddTrack(caller->CreateAudioTrack("audio_track"),
{"audio_stream"}));
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
auto& add_track_event = callee->observer()->add_track_events_[0];
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_EQ(add_track_event.streams.size(), 1u);
EXPECT_EQ("audio_stream", add_track_event.streams[0]->label());
EXPECT_TRUE(add_track_event.streams[0]->FindAudioTrack("audio_track"));
EXPECT_EQ(add_track_event.streams, add_track_event.receiver->streams());
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
}
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
TEST_F(PeerConnectionRtpCallbacksTest,
RemoveTrackWithoutStreamFiresOnRemoveTrack) {
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
auto sender = caller->AddTrack(caller->CreateAudioTrack("audio_track"), {});
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
EXPECT_TRUE(caller->pc()->RemoveTrack(sender));
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
EXPECT_EQ(callee->observer()->GetAddTrackReceivers(),
callee->observer()->remove_track_events_);
}
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
TEST_F(PeerConnectionRtpCallbacksTest,
RemoveTrackWithStreamFiresOnRemoveTrack) {
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
auto sender = caller->AddTrack(caller->CreateAudioTrack("audio_track"),
{"audio_stream"});
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
EXPECT_TRUE(caller->pc()->RemoveTrack(sender));
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
EXPECT_EQ(callee->observer()->GetAddTrackReceivers(),
callee->observer()->remove_track_events_);
}
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
TEST_F(PeerConnectionRtpCallbacksTest,
RemoveTrackWithSharedStreamFiresOnRemoveTrack) {
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
const char kSharedStreamLabel[] = "shared_audio_stream";
auto sender1 = caller->AddTrack(caller->CreateAudioTrack("audio_track1"),
{kSharedStreamLabel});
auto sender2 = caller->AddTrack(caller->CreateAudioTrack("audio_track2"),
{kSharedStreamLabel});
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_EQ(callee->observer()->add_track_events_.size(), 2u);
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
// Remove "audio_track1".
EXPECT_TRUE(caller->pc()->RemoveTrack(sender1));
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
ASSERT_EQ(callee->observer()->add_track_events_.size(), 2u);
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
EXPECT_EQ(
Reland "Add AddTransceiver and GetTransceivers to PeerConnection" This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80. Original change's description: > Revert "Add AddTransceiver and GetTransceivers to PeerConnection" > > This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a. > > Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout > > Original change's description: > > Add AddTransceiver and GetTransceivers to PeerConnection > > > > WebRTC 1.0 has added the transceiver API to PeerConnection. This > > is the first step towards exposing this to WebRTC consumers. For > > now, transceivers can be added and fetched but there is not yet > > support for creating offers/answers or setting local/remote > > descriptions. That support ("Unified Plan") will be added in > > follow-up CLs. > > > > The transceiver API is currently only available if the application > > opts in by specifying the kUnifiedPlan SDP semantics when creating > > the PeerConnection. > > > > Bug: webrtc:7600 > > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e > > Reviewed-on: https://webrtc-review.googlesource.com/23880 > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20896} > > TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org > > Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7600 > Reviewed-on: https://webrtc-review.googlesource.com/26400 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20897} TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7600 Reviewed-on: https://webrtc-review.googlesource.com/26401 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20898}
2017-11-27 13:01:52 -08:00
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>{
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
callee->observer()->add_track_events_[0].receiver},
callee->observer()->remove_track_events_);
// Remove "audio_track2".
EXPECT_TRUE(caller->pc()->RemoveTrack(sender2));
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
ASSERT_EQ(callee->observer()->add_track_events_.size(), 2u);
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
EXPECT_EQ(callee->observer()->GetAddTrackReceivers(),
callee->observer()->remove_track_events_);
}
// Tests that setting a remote description with sending transceivers will fire
// the OnTrack callback for each transceiver and setting a remote description
// with receive only transceivers will not call OnTrack.
TEST_F(PeerConnectionRtpCallbacksTest, UnifiedPlanAddTransceiverCallsOnTrack) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto callee = CreatePeerConnectionWithUnifiedPlan();
auto audio_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
auto video_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
ASSERT_EQ(0u, caller->observer()->on_track_transceivers_.size());
ASSERT_EQ(2u, callee->observer()->on_track_transceivers_.size());
EXPECT_EQ(audio_transceiver->mid(),
callee->pc()->GetTransceivers()[0]->mid());
EXPECT_EQ(video_transceiver->mid(),
callee->pc()->GetTransceivers()[1]->mid());
}
// Test that doing additional offer/answer exchanges with no changes to tracks
// will cause no additional OnTrack calls after the tracks have been negotiated.
TEST_F(PeerConnectionRtpCallbacksTest, UnifiedPlanReofferDoesNotCallOnTrack) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto callee = CreatePeerConnectionWithUnifiedPlan();
caller->AddAudioTrack("audio");
callee->AddAudioTrack("audio");
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
EXPECT_EQ(1u, caller->observer()->on_track_transceivers_.size());
EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
// If caller reoffers with no changes expect no additional OnTrack calls.
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
EXPECT_EQ(1u, caller->observer()->on_track_transceivers_.size());
EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
// Also if callee reoffers with no changes expect no additional OnTrack calls.
ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get()));
EXPECT_EQ(1u, caller->observer()->on_track_transceivers_.size());
EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
}
// Test that OnTrack is called when the transceiver direction changes to send
// the track.
TEST_F(PeerConnectionRtpCallbacksTest, UnifiedPlanSetDirectionCallsOnTrack) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto callee = CreatePeerConnectionWithUnifiedPlan();
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
transceiver->SetDirection(RtpTransceiverDirection::kInactive);
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
EXPECT_EQ(0u, callee->observer()->on_track_transceivers_.size());
transceiver->SetDirection(RtpTransceiverDirection::kSendOnly);
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
// If the direction changes but it is still receiving on the remote side, then
// OnTrack should not be fired again.
transceiver->SetDirection(RtpTransceiverDirection::kSendRecv);
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
}
// Test that OnTrack is called twice when a sendrecv call is started, the callee
// changes the direction to inactive, then changes it back to sendrecv.
TEST_F(PeerConnectionRtpCallbacksTest,
UnifiedPlanSetDirectionHoldCallsOnTrackTwice) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto callee = CreatePeerConnectionWithUnifiedPlan();
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
// Put the call on hold by no longer receiving the track.
callee->pc()->GetTransceivers()[0]->SetDirection(
RtpTransceiverDirection::kInactive);
ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get()));
EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
// Resume the call by changing the direction to recvonly. This should call
// OnTrack again on the callee side.
callee->pc()->GetTransceivers()[0]->SetDirection(
RtpTransceiverDirection::kRecvOnly);
ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get()));
EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
EXPECT_EQ(2u, callee->observer()->on_track_transceivers_.size());
}
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
// These tests examine the state of the peer connection as a result of
// performing SetRemoteDescription().
class PeerConnectionRtpObserverTest : public PeerConnectionRtpTest {};
TEST_F(PeerConnectionRtpObserverTest, AddSenderWithoutStreamAddsReceiver) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
ASSERT_TRUE(caller->AddTrack(caller->CreateAudioTrack("audio_track"), {}));
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
EXPECT_EQ(callee->pc()->GetReceivers().size(), 1u);
auto receiver_added = callee->pc()->GetReceivers()[0];
EXPECT_EQ("audio_track", receiver_added->track()->id());
// TODO(hbos): When "no stream" is handled correctly we would expect
// |receiver_added->streams()| to be empty. https://crbug.com/webrtc/7933
EXPECT_EQ(receiver_added->streams().size(), 1u);
EXPECT_TRUE(receiver_added->streams()[0]->FindAudioTrack("audio_track"));
}
TEST_F(PeerConnectionRtpObserverTest, AddSenderWithStreamAddsReceiver) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
ASSERT_TRUE(caller->AddTrack(caller->CreateAudioTrack("audio_track"),
{"audio_stream"}));
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
EXPECT_EQ(callee->pc()->GetReceivers().size(), 1u);
auto receiver_added = callee->pc()->GetReceivers()[0];
EXPECT_EQ("audio_track", receiver_added->track()->id());
EXPECT_EQ(receiver_added->streams().size(), 1u);
EXPECT_EQ("audio_stream", receiver_added->streams()[0]->label());
EXPECT_TRUE(receiver_added->streams()[0]->FindAudioTrack("audio_track"));
}
TEST_F(PeerConnectionRtpObserverTest,
RemoveSenderWithoutStreamRemovesReceiver) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
auto sender = caller->AddTrack(caller->CreateAudioTrack("audio_track"), {});
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_TRUE(sender);
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
ASSERT_EQ(callee->pc()->GetReceivers().size(), 1u);
auto receiver = callee->pc()->GetReceivers()[0];
ASSERT_TRUE(caller->pc()->RemoveTrack(sender));
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
// TODO(hbos): When we implement Unified Plan, receivers will not be removed.
// Instead, the transceiver owning the receiver will become inactive.
EXPECT_EQ(callee->pc()->GetReceivers().size(), 0u);
}
TEST_F(PeerConnectionRtpObserverTest, RemoveSenderWithStreamRemovesReceiver) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
auto sender = caller->AddTrack(caller->CreateAudioTrack("audio_track"),
{"audio_stream"});
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_TRUE(sender);
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
ASSERT_EQ(callee->pc()->GetReceivers().size(), 1u);
auto receiver = callee->pc()->GetReceivers()[0];
ASSERT_TRUE(caller->pc()->RemoveTrack(sender));
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
// TODO(hbos): When we implement Unified Plan, receivers will not be removed.
// Instead, the transceiver owning the receiver will become inactive.
EXPECT_EQ(callee->pc()->GetReceivers().size(), 0u);
}
TEST_F(PeerConnectionRtpObserverTest,
RemoveSenderWithSharedStreamRemovesReceiver) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
const char kSharedStreamLabel[] = "shared_audio_stream";
auto sender1 = caller->AddTrack(caller->CreateAudioTrack("audio_track1"),
{kSharedStreamLabel});
auto sender2 = caller->AddTrack(caller->CreateAudioTrack("audio_track2"),
{kSharedStreamLabel});
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
ASSERT_EQ(callee->pc()->GetReceivers().size(), 2u);
rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver1;
rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver2;
if (callee->pc()->GetReceivers()[0]->track()->id() == "audio_track1") {
receiver1 = callee->pc()->GetReceivers()[0];
receiver2 = callee->pc()->GetReceivers()[1];
} else {
receiver1 = callee->pc()->GetReceivers()[1];
receiver2 = callee->pc()->GetReceivers()[0];
}
EXPECT_EQ("audio_track1", receiver1->track()->id());
EXPECT_EQ("audio_track2", receiver2->track()->id());
// Remove "audio_track1".
EXPECT_TRUE(caller->pc()->RemoveTrack(sender1));
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
// Only |receiver2| should remain.
// TODO(hbos): When we implement Unified Plan, receivers will not be removed.
// Instead, the transceiver owning the receiver will become inactive.
EXPECT_EQ(
std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>>{receiver2},
callee->pc()->GetReceivers());
// Remove "audio_track2".
EXPECT_TRUE(caller->pc()->RemoveTrack(sender2));
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
static_cast<webrtc::RTCError*>(nullptr)));
// TODO(hbos): When we implement Unified Plan, receivers will not be removed.
// Instead, the transceiver owning the receiver will become inactive.
EXPECT_EQ(callee->pc()->GetReceivers().size(), 0u);
}
// Invokes SetRemoteDescription() twice in a row without synchronizing the two
// calls and examine the state of the peer connection inside the callbacks to
// ensure that the second call does not occur prematurely, contaminating the
// state of the peer connection of the first callback.
TEST_F(PeerConnectionRtpObserverTest,
StatesCorrelateWithSetRemoteDescriptionCall) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
// Create SDP for adding a track and for removing it. This will be used in the
// first and second SetRemoteDescription() calls.
auto sender = caller->AddTrack(caller->CreateAudioTrack("audio_track"), {});
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
auto srd1_sdp = caller->CreateOfferAndSetAsLocal();
EXPECT_TRUE(caller->pc()->RemoveTrack(sender));
auto srd2_sdp = caller->CreateOfferAndSetAsLocal();
// In the first SetRemoteDescription() callback, check that we have a
// receiver for the track.
auto pc = callee->pc();
bool srd1_callback_called = false;
auto srd1_callback = [&srd1_callback_called, &pc]() {
EXPECT_EQ(pc->GetReceivers().size(), 1u);
srd1_callback_called = true;
};
// In the second SetRemoteDescription() callback, check that the receiver has
// been removed.
// TODO(hbos): When we implement Unified Plan, receivers will not be removed.
// Instead, the transceiver owning the receiver will become inactive.
// https://crbug.com/webrtc/7600
bool srd2_callback_called = false;
auto srd2_callback = [&srd2_callback_called, &pc]() {
EXPECT_TRUE(pc->GetReceivers().empty());
srd2_callback_called = true;
};
// Invoke SetRemoteDescription() twice in a row without synchronizing the two
// calls. The callbacks verify that the two calls are synchronized, as in, the
// effects of the second SetRemoteDescription() call must not have happened by
// the time the first callback is invoked. If it has then the receiver that is
// added as a result of the first SetRemoteDescription() call will already
// have been removed as a result of the second SetRemoteDescription() call
// when the first callback is invoked.
callee->pc()->SetRemoteDescription(
std::move(srd1_sdp),
new OnSuccessObserver<decltype(srd1_callback)>(srd1_callback));
callee->pc()->SetRemoteDescription(
std::move(srd2_sdp),
new OnSuccessObserver<decltype(srd2_callback)>(srd2_callback));
EXPECT_TRUE_WAIT(srd1_callback_called, kDefaultTimeout);
EXPECT_TRUE_WAIT(srd2_callback_called, kDefaultTimeout);
}
// Tests for the legacy SetRemoteDescription() function signature.
class PeerConnectionRtpLegacyObserverTest : public PeerConnectionRtpTest {};
// Sanity test making sure the callback is invoked.
TEST_F(PeerConnectionRtpLegacyObserverTest, OnSuccess) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
std::string error;
ASSERT_TRUE(
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(), &error));
}
// Verifies legacy behavior: The observer is not called if if the peer
// connection is destroyed because the asynchronous callback is executed in the
// peer connection's message handler.
TEST_F(PeerConnectionRtpLegacyObserverTest,
ObserverNotCalledIfPeerConnectionDereferenced) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
rtc::scoped_refptr<webrtc::MockSetSessionDescriptionObserver> observer =
new rtc::RefCountedObject<webrtc::MockSetSessionDescriptionObserver>();
auto offer = caller->CreateOfferAndSetAsLocal();
callee->pc()->SetRemoteDescription(observer, offer.release());
callee = nullptr;
rtc::Thread::Current()->ProcessMessages(0);
EXPECT_FALSE(observer->called());
}
// RtpTransceiver Tests.
Reland "Add AddTransceiver and GetTransceivers to PeerConnection" This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80. Original change's description: > Revert "Add AddTransceiver and GetTransceivers to PeerConnection" > > This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a. > > Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout > > Original change's description: > > Add AddTransceiver and GetTransceivers to PeerConnection > > > > WebRTC 1.0 has added the transceiver API to PeerConnection. This > > is the first step towards exposing this to WebRTC consumers. For > > now, transceivers can be added and fetched but there is not yet > > support for creating offers/answers or setting local/remote > > descriptions. That support ("Unified Plan") will be added in > > follow-up CLs. > > > > The transceiver API is currently only available if the application > > opts in by specifying the kUnifiedPlan SDP semantics when creating > > the PeerConnection. > > > > Bug: webrtc:7600 > > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e > > Reviewed-on: https://webrtc-review.googlesource.com/23880 > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20896} > > TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org > > Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7600 > Reviewed-on: https://webrtc-review.googlesource.com/26400 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20897} TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7600 Reviewed-on: https://webrtc-review.googlesource.com/26401 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20898}
2017-11-27 13:01:52 -08:00
// Test that by default there are no transceivers with Unified Plan.
TEST_F(PeerConnectionRtpTest, PeerConnectionHasNoTransceivers) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
EXPECT_THAT(caller->pc()->GetTransceivers(), ElementsAre());
}
// Test that a transceiver created with the audio kind has the correct initial
// properties.
TEST_F(PeerConnectionRtpTest, AddTransceiverHasCorrectInitProperties) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
EXPECT_EQ(rtc::nullopt, transceiver->mid());
EXPECT_FALSE(transceiver->stopped());
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceiver->direction());
EXPECT_EQ(rtc::nullopt, transceiver->current_direction());
}
// Test that adding a transceiver with the audio kind creates an audio sender
// and audio receiver with the receiver having a live audio track.
TEST_F(PeerConnectionRtpTest,
AddAudioTransceiverCreatesAudioSenderAndReceiver) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, transceiver->media_type());
Reland "Add AddTransceiver and GetTransceivers to PeerConnection" This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80. Original change's description: > Revert "Add AddTransceiver and GetTransceivers to PeerConnection" > > This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a. > > Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout > > Original change's description: > > Add AddTransceiver and GetTransceivers to PeerConnection > > > > WebRTC 1.0 has added the transceiver API to PeerConnection. This > > is the first step towards exposing this to WebRTC consumers. For > > now, transceivers can be added and fetched but there is not yet > > support for creating offers/answers or setting local/remote > > descriptions. That support ("Unified Plan") will be added in > > follow-up CLs. > > > > The transceiver API is currently only available if the application > > opts in by specifying the kUnifiedPlan SDP semantics when creating > > the PeerConnection. > > > > Bug: webrtc:7600 > > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e > > Reviewed-on: https://webrtc-review.googlesource.com/23880 > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20896} > > TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org > > Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7600 > Reviewed-on: https://webrtc-review.googlesource.com/26400 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20897} TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7600 Reviewed-on: https://webrtc-review.googlesource.com/26401 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20898}
2017-11-27 13:01:52 -08:00
ASSERT_TRUE(transceiver->sender());
EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, transceiver->sender()->media_type());
ASSERT_TRUE(transceiver->receiver());
EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, transceiver->receiver()->media_type());
auto track = transceiver->receiver()->track();
ASSERT_TRUE(track);
EXPECT_EQ(MediaStreamTrackInterface::kAudioKind, track->kind());
EXPECT_EQ(MediaStreamTrackInterface::TrackState::kLive, track->state());
}
// Test that adding a transceiver with the video kind creates an video sender
// and video receiver with the receiver having a live video track.
TEST_F(PeerConnectionRtpTest,
AddAudioTransceiverCreatesVideoSenderAndReceiver) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->media_type());
Reland "Add AddTransceiver and GetTransceivers to PeerConnection" This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80. Original change's description: > Revert "Add AddTransceiver and GetTransceivers to PeerConnection" > > This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a. > > Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout > > Original change's description: > > Add AddTransceiver and GetTransceivers to PeerConnection > > > > WebRTC 1.0 has added the transceiver API to PeerConnection. This > > is the first step towards exposing this to WebRTC consumers. For > > now, transceivers can be added and fetched but there is not yet > > support for creating offers/answers or setting local/remote > > descriptions. That support ("Unified Plan") will be added in > > follow-up CLs. > > > > The transceiver API is currently only available if the application > > opts in by specifying the kUnifiedPlan SDP semantics when creating > > the PeerConnection. > > > > Bug: webrtc:7600 > > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e > > Reviewed-on: https://webrtc-review.googlesource.com/23880 > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20896} > > TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org > > Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7600 > Reviewed-on: https://webrtc-review.googlesource.com/26400 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20897} TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7600 Reviewed-on: https://webrtc-review.googlesource.com/26401 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20898}
2017-11-27 13:01:52 -08:00
ASSERT_TRUE(transceiver->sender());
EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->sender()->media_type());
ASSERT_TRUE(transceiver->receiver());
EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->receiver()->media_type());
auto track = transceiver->receiver()->track();
ASSERT_TRUE(track);
EXPECT_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());
EXPECT_EQ(MediaStreamTrackInterface::TrackState::kLive, track->state());
}
// Test that after a call to AddTransceiver, the transceiver shows in
// GetTransceivers(), the transceiver's sender shows in GetSenders(), and the
// transceiver's receiver shows in GetReceivers().
TEST_F(PeerConnectionRtpTest, AddTransceiverShowsInLists) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
EXPECT_EQ(
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>{transceiver},
caller->pc()->GetTransceivers());
EXPECT_EQ(
std::vector<rtc::scoped_refptr<RtpSenderInterface>>{
transceiver->sender()},
caller->pc()->GetSenders());
EXPECT_EQ(
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>{
transceiver->receiver()},
caller->pc()->GetReceivers());
}
// Test that the direction passed in through the AddTransceiver init parameter
// is set in the returned transceiver.
TEST_F(PeerConnectionRtpTest, AddTransceiverWithDirectionIsReflected) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
RtpTransceiverInit init;
init.direction = RtpTransceiverDirection::kSendOnly;
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
EXPECT_EQ(RtpTransceiverDirection::kSendOnly, transceiver->direction());
}
// Test that calling AddTransceiver with a track creates a transceiver which has
// its sender's track set to the passed-in track.
TEST_F(PeerConnectionRtpTest, AddTransceiverWithTrackCreatesSenderWithTrack) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto audio_track = caller->CreateAudioTrack("audio track");
auto transceiver = caller->AddTransceiver(audio_track);
auto sender = transceiver->sender();
ASSERT_TRUE(sender->track());
EXPECT_EQ(audio_track, sender->track());
auto receiver = transceiver->receiver();
ASSERT_TRUE(receiver->track());
EXPECT_EQ(MediaStreamTrackInterface::kAudioKind, receiver->track()->kind());
EXPECT_EQ(MediaStreamTrackInterface::TrackState::kLive,
receiver->track()->state());
}
// Test that calling AddTransceiver twice with the same track creates distinct
// transceivers, senders with the same track.
TEST_F(PeerConnectionRtpTest,
AddTransceiverTwiceWithSameTrackCreatesMultipleTransceivers) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto audio_track = caller->CreateAudioTrack("audio track");
auto transceiver1 = caller->AddTransceiver(audio_track);
auto transceiver2 = caller->AddTransceiver(audio_track);
EXPECT_NE(transceiver1, transceiver2);
auto sender1 = transceiver1->sender();
auto sender2 = transceiver2->sender();
EXPECT_NE(sender1, sender2);
EXPECT_EQ(audio_track, sender1->track());
EXPECT_EQ(audio_track, sender2->track());
EXPECT_THAT(caller->pc()->GetTransceivers(),
UnorderedElementsAre(transceiver1, transceiver2));
EXPECT_THAT(caller->pc()->GetSenders(),
UnorderedElementsAre(sender1, sender2));
}
// RtpTransceiver error handling tests.
TEST_F(PeerConnectionRtpTest, AddTransceiverWithInvalidKindReturnsError) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto result = caller->pc()->AddTransceiver(cricket::MEDIA_TYPE_DATA);
EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, result.error().type());
}
TEST_F(PeerConnectionRtpTest, UnifiedPlanCanClosePeerConnection) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
caller->pc()->Close();
}
// Unified Plan AddTrack tests.
class PeerConnectionRtpUnifiedPlanTest : public PeerConnectionRtpTest {};
// Test that adding an audio track creates a new audio RtpSender with the given
// track.
TEST_F(PeerConnectionRtpUnifiedPlanTest, AddAudioTrackCreatesAudioSender) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto audio_track = caller->CreateAudioTrack("a");
auto sender = caller->AddTrack(audio_track);
ASSERT_TRUE(sender);
EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, sender->media_type());
EXPECT_EQ(audio_track, sender->track());
}
// Test that adding a video track creates a new video RtpSender with the given
// track.
TEST_F(PeerConnectionRtpUnifiedPlanTest, AddVideoTrackCreatesVideoSender) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto video_track = caller->CreateVideoTrack("a");
auto sender = caller->AddTrack(video_track);
ASSERT_TRUE(sender);
EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, sender->media_type());
EXPECT_EQ(video_track, sender->track());
}
// Test that adding a track to a new PeerConnection creates an RtpTransceiver
// with the sender that AddTrack returns and in the sendrecv direction.
TEST_F(PeerConnectionRtpUnifiedPlanTest, AddFirstTrackCreatesTransceiver) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto sender = caller->AddAudioTrack("a");
ASSERT_TRUE(sender);
auto transceivers = caller->pc()->GetTransceivers();
ASSERT_EQ(1u, transceivers.size());
EXPECT_EQ(sender, transceivers[0]->sender());
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceivers[0]->direction());
}
// Test that if a transceiver of the same type but no track had been added to
// the PeerConnection and later a call to AddTrack is made, the resulting sender
// is the transceiver's sender and the sender's track is the newly-added track.
TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackReusesTransceiver) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
auto audio_track = caller->CreateAudioTrack("a");
auto sender = caller->AddTrack(audio_track);
ASSERT_TRUE(sender);
auto transceivers = caller->pc()->GetTransceivers();
ASSERT_EQ(1u, transceivers.size());
EXPECT_EQ(transceiver, transceivers[0]);
EXPECT_EQ(sender, transceiver->sender());
EXPECT_EQ(audio_track, sender->track());
}
// Test that adding two tracks to a new PeerConnection creates two
// RtpTransceivers in the same order.
TEST_F(PeerConnectionRtpUnifiedPlanTest, TwoAddTrackCreatesTwoTransceivers) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto sender1 = caller->AddAudioTrack("a");
auto sender2 = caller->AddVideoTrack("v");
ASSERT_TRUE(sender2);
auto transceivers = caller->pc()->GetTransceivers();
ASSERT_EQ(2u, transceivers.size());
EXPECT_EQ(sender1, transceivers[0]->sender());
EXPECT_EQ(sender2, transceivers[1]->sender());
}
// Test that if there are multiple transceivers with no sending track then a
// later call to AddTrack will use the one of the same type as the newly-added
// track.
TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackReusesTransceiverOfType) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto audio_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
auto video_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
auto sender = caller->AddVideoTrack("v");
ASSERT_EQ(2u, caller->pc()->GetTransceivers().size());
EXPECT_NE(sender, audio_transceiver->sender());
EXPECT_EQ(sender, video_transceiver->sender());
}
// Test that if the only transceivers that do not have a sending track have a
// different type from the added track, then AddTrack will create a new
// transceiver for the track.
TEST_F(PeerConnectionRtpUnifiedPlanTest,
AddTrackDoesNotReuseTransceiverOfWrongType) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
auto sender = caller->AddVideoTrack("v");
auto transceivers = caller->pc()->GetTransceivers();
ASSERT_EQ(2u, transceivers.size());
EXPECT_NE(sender, transceivers[0]->sender());
EXPECT_EQ(sender, transceivers[1]->sender());
}
// Test that the first available transceiver is reused by AddTrack when multiple
// are available.
TEST_F(PeerConnectionRtpUnifiedPlanTest,
AddTrackReusesFirstMatchingTransceiver) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
auto sender = caller->AddAudioTrack("a");
auto transceivers = caller->pc()->GetTransceivers();
ASSERT_EQ(2u, transceivers.size());
EXPECT_EQ(sender, transceivers[0]->sender());
EXPECT_NE(sender, transceivers[1]->sender());
}
// Test that a call to AddTrack that reuses a transceiver will change the
// direction from inactive to sendonly.
TEST_F(PeerConnectionRtpUnifiedPlanTest,
AddTrackChangesDirectionFromInactiveToSendOnly) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
RtpTransceiverInit init;
init.direction = RtpTransceiverDirection::kInactive;
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
caller->observer()->clear_negotiation_needed();
ASSERT_TRUE(caller->AddAudioTrack("a"));
EXPECT_TRUE(caller->observer()->negotiation_needed());
EXPECT_EQ(RtpTransceiverDirection::kSendOnly, transceiver->direction());
}
// Test that a call to AddTrack that reuses a transceiver will change the
// direction from recvonly to sendrecv.
TEST_F(PeerConnectionRtpUnifiedPlanTest,
AddTrackChangesDirectionFromRecvOnlyToSendRecv) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
RtpTransceiverInit init;
init.direction = RtpTransceiverDirection::kRecvOnly;
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
caller->observer()->clear_negotiation_needed();
ASSERT_TRUE(caller->AddAudioTrack("a"));
EXPECT_TRUE(caller->observer()->negotiation_needed());
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceiver->direction());
}
TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackCreatesSenderWithTrackId) {
const std::string kTrackId = "audio_track";
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto audio_track = caller->CreateAudioTrack(kTrackId);
auto sender = caller->AddTrack(audio_track);
EXPECT_EQ(kTrackId, sender->id());
}
// Unified Plan AddTrack error handling.
TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackErrorIfClosed) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto audio_track = caller->CreateAudioTrack("a");
caller->pc()->Close();
caller->observer()->clear_negotiation_needed();
auto result = caller->pc()
->AddTrack(audio_track, std::vector<std::string>());
EXPECT_EQ(RTCErrorType::INVALID_STATE, result.error().type());
EXPECT_FALSE(caller->observer()->negotiation_needed());
}
TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackErrorIfTrackAlreadyHasSender) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto audio_track = caller->CreateAudioTrack("a");
ASSERT_TRUE(caller->AddTrack(audio_track));
caller->observer()->clear_negotiation_needed();
auto result = caller->pc()
->AddTrack(audio_track, std::vector<std::string>());
EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, result.error().type());
EXPECT_FALSE(caller->observer()->negotiation_needed());
}
// Unified Plan RemoveTrack tests.
// Test that calling RemoveTrack on a sender with a previously-added track
// clears the sender's track.
TEST_F(PeerConnectionRtpUnifiedPlanTest, RemoveTrackClearsSenderTrack) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto sender = caller->AddAudioTrack("a");
ASSERT_TRUE(caller->pc()->RemoveTrack(sender));
EXPECT_FALSE(sender->track());
}
// Test that calling RemoveTrack on a sender where the transceiver is configured
// in the sendrecv direction changes the transceiver's direction to recvonly.
TEST_F(PeerConnectionRtpUnifiedPlanTest,
RemoveTrackChangesDirectionFromSendRecvToRecvOnly) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
RtpTransceiverInit init;
init.direction = RtpTransceiverDirection::kSendRecv;
auto transceiver =
caller->AddTransceiver(caller->CreateAudioTrack("a"), init);
caller->observer()->clear_negotiation_needed();
ASSERT_TRUE(caller->pc()->RemoveTrack(transceiver->sender()));
EXPECT_TRUE(caller->observer()->negotiation_needed());
EXPECT_EQ(RtpTransceiverDirection::kRecvOnly, transceiver->direction());
EXPECT_TRUE(caller->observer()->renegotiation_needed_);
}
// Test that calling RemoveTrack on a sender where the transceiver is configured
// in the sendonly direction changes the transceiver's direction to inactive.
TEST_F(PeerConnectionRtpUnifiedPlanTest,
RemoveTrackChangesDirectionFromSendOnlyToInactive) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
RtpTransceiverInit init;
init.direction = RtpTransceiverDirection::kSendOnly;
auto transceiver =
caller->AddTransceiver(caller->CreateAudioTrack("a"), init);
caller->observer()->clear_negotiation_needed();
ASSERT_TRUE(caller->pc()->RemoveTrack(transceiver->sender()));
EXPECT_TRUE(caller->observer()->negotiation_needed());
EXPECT_EQ(RtpTransceiverDirection::kInactive, transceiver->direction());
}
// Test that calling RemoveTrack with a sender that has a null track results in
// no change in state.
TEST_F(PeerConnectionRtpUnifiedPlanTest, RemoveTrackWithNullSenderTrackIsNoOp) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto sender = caller->AddAudioTrack("a");
auto transceiver = caller->pc()->GetTransceivers()[0];
ASSERT_TRUE(sender->SetTrack(nullptr));
caller->observer()->clear_negotiation_needed();
ASSERT_TRUE(caller->pc()->RemoveTrack(sender));
EXPECT_FALSE(caller->observer()->negotiation_needed());
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceiver->direction());
}
// Unified Plan RemoveTrack error handling.
TEST_F(PeerConnectionRtpUnifiedPlanTest, RemoveTrackErrorIfClosed) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto sender = caller->AddAudioTrack("a");
caller->pc()->Close();
caller->observer()->clear_negotiation_needed();
EXPECT_FALSE(caller->pc()->RemoveTrack(sender));
EXPECT_FALSE(caller->observer()->negotiation_needed());
}
TEST_F(PeerConnectionRtpUnifiedPlanTest,
RemoveTrackNoErrorIfTrackAlreadyRemoved) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
auto sender = caller->AddAudioTrack("a");
ASSERT_TRUE(caller->pc()->RemoveTrack(sender));
caller->observer()->clear_negotiation_needed();
EXPECT_TRUE(caller->pc()->RemoveTrack(sender));
EXPECT_FALSE(caller->observer()->negotiation_needed());
}
// Test MSID signaling between Unified Plan and Plan B endpoints. There are two
// options for this kind of signaling: media section based (a=msid) and ssrc
// based (a=ssrc MSID). While JSEP only specifies media section MSID signaling,
// we want to ensure compatibility with older Plan B endpoints that might expect
// ssrc based MSID signaling. Thus we test here that Unified Plan offers both
// types but answers with the same type as the offer.
class PeerConnectionMsidSignalingTest : public PeerConnectionRtpTest {};
TEST_F(PeerConnectionMsidSignalingTest, UnifiedPlanTalkingToOurself) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
caller->AddAudioTrack("caller_audio");
auto callee = CreatePeerConnectionWithUnifiedPlan();
callee->AddAudioTrack("callee_audio");
auto caller_observer =
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
caller->pc()->RegisterUMAObserver(caller_observer);
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
// Offer should have had both a=msid and a=ssrc MSID lines.
auto* offer = callee->pc()->remote_description();
EXPECT_EQ((cricket::kMsidSignalingMediaSection |
cricket::kMsidSignalingSsrcAttribute),
offer->description()->msid_signaling());
// Answer should have had only a=msid lines.
auto* answer = caller->pc()->remote_description();
EXPECT_EQ(cricket::kMsidSignalingMediaSection,
answer->description()->msid_signaling());
// Check that this is counted correctly
EXPECT_EQ(1, caller_observer->GetEnumCounter(
webrtc::kEnumCounterSdpSemanticNegotiated,
webrtc::kSdpSemanticNegotiatedUnifiedPlan));
EXPECT_EQ(0, caller_observer->GetEnumCounter(
webrtc::kEnumCounterSdpSemanticNegotiated,
webrtc::kSdpSemanticNegotiatedNone));
EXPECT_EQ(0, caller_observer->GetEnumCounter(
webrtc::kEnumCounterSdpSemanticNegotiated,
webrtc::kSdpSemanticNegotiatedPlanB));
EXPECT_EQ(0, caller_observer->GetEnumCounter(
webrtc::kEnumCounterSdpSemanticNegotiated,
webrtc::kSdpSemanticNegotiatedMixed));
}
TEST_F(PeerConnectionMsidSignalingTest, PlanBOfferToUnifiedPlanAnswer) {
auto caller = CreatePeerConnectionWithPlanB();
caller->AddAudioTrack("caller_audio");
auto callee = CreatePeerConnectionWithUnifiedPlan();
callee->AddAudioTrack("callee_audio");
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
// Offer should have only a=ssrc MSID lines.
auto* offer = callee->pc()->remote_description();
EXPECT_EQ(cricket::kMsidSignalingSsrcAttribute,
offer->description()->msid_signaling());
// Answer should have only a=ssrc MSID lines to match the offer.
auto* answer = caller->pc()->remote_description();
EXPECT_EQ(cricket::kMsidSignalingSsrcAttribute,
answer->description()->msid_signaling());
}
TEST_F(PeerConnectionMsidSignalingTest, PureUnifiedPlanToUs) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
caller->AddAudioTrack("caller_audio");
auto callee = CreatePeerConnectionWithUnifiedPlan();
callee->AddAudioTrack("callee_audio");
auto offer = caller->CreateOffer();
// Simulate a pure Unified Plan offerer by setting the MSID signaling to media
// section only.
offer->description()->set_msid_signaling(cricket::kMsidSignalingMediaSection);
ASSERT_TRUE(
caller->SetLocalDescription(CloneSessionDescription(offer.get())));
ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
// Answer should have only a=msid to match the offer.
auto answer = callee->CreateAnswer();
EXPECT_EQ(cricket::kMsidSignalingMediaSection,
answer->description()->msid_signaling());
}
// Sender setups in a call.
class PeerConnectionSenderTest : public PeerConnectionRtpTest {};
TEST_F(PeerConnectionSenderTest, CreateTwoSendersWithSameTrack) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
auto track = caller->CreateAudioTrack("audio_track");
auto sender1 = caller->AddTrack(track);
ASSERT_TRUE(sender1);
// We need to temporarily reset the track for the subsequent AddTrack() to
// succeed.
EXPECT_TRUE(sender1->SetTrack(nullptr));
auto sender2 = caller->AddTrack(track);
EXPECT_TRUE(sender2);
EXPECT_TRUE(sender1->SetTrack(track));
// TODO(hbos): When https://crbug.com/webrtc/8734 is resolved, this should
// return true, and doing |callee->SetRemoteDescription()| should work.
EXPECT_FALSE(caller->CreateOfferAndSetAsLocal());
}
Reland "Add AddTransceiver and GetTransceivers to PeerConnection" This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80. Original change's description: > Revert "Add AddTransceiver and GetTransceivers to PeerConnection" > > This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a. > > Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout > > Original change's description: > > Add AddTransceiver and GetTransceivers to PeerConnection > > > > WebRTC 1.0 has added the transceiver API to PeerConnection. This > > is the first step towards exposing this to WebRTC consumers. For > > now, transceivers can be added and fetched but there is not yet > > support for creating offers/answers or setting local/remote > > descriptions. That support ("Unified Plan") will be added in > > follow-up CLs. > > > > The transceiver API is currently only available if the application > > opts in by specifying the kUnifiedPlan SDP semantics when creating > > the PeerConnection. > > > > Bug: webrtc:7600 > > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e > > Reviewed-on: https://webrtc-review.googlesource.com/23880 > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20896} > > TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org > > Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7600 > Reviewed-on: https://webrtc-review.googlesource.com/26400 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20897} TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7600 Reviewed-on: https://webrtc-review.googlesource.com/26401 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20898}
2017-11-27 13:01:52 -08:00
} // namespace webrtc