webrtc_m130/pc/peer_connection_wrapper.cc

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/*
* 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 "pc/peer_connection_wrapper.h"
#include <stdint.h>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "api/function_view.h"
#include "api/set_remote_description_observer_interface.h"
#include "pc/sdp_utils.h"
#include "pc/test/fake_video_track_source.h"
#include "rtc_base/checks.h"
#include "rtc_base/gunit.h"
#include "rtc_base/logging.h"
#include "rtc_base/ref_counted_object.h"
#include "test/gtest.h"
namespace webrtc {
using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
namespace {
const uint32_t kDefaultTimeout = 10000U;
}
PeerConnectionWrapper::PeerConnectionWrapper(
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory,
rtc::scoped_refptr<PeerConnectionInterface> pc,
std::unique_ptr<MockPeerConnectionObserver> observer)
: pc_factory_(std::move(pc_factory)),
observer_(std::move(observer)),
pc_(std::move(pc)) {
RTC_DCHECK(pc_factory_);
RTC_DCHECK(pc_);
RTC_DCHECK(observer_);
observer_->SetPeerConnectionInterface(pc_.get());
}
PeerConnectionWrapper::~PeerConnectionWrapper() = default;
PeerConnectionFactoryInterface* PeerConnectionWrapper::pc_factory() {
return pc_factory_.get();
}
PeerConnectionInterface* PeerConnectionWrapper::pc() {
return pc_.get();
}
MockPeerConnectionObserver* PeerConnectionWrapper::observer() {
return observer_.get();
}
std::unique_ptr<SessionDescriptionInterface>
PeerConnectionWrapper::CreateOffer() {
return CreateOffer(RTCOfferAnswerOptions());
}
std::unique_ptr<SessionDescriptionInterface> PeerConnectionWrapper::CreateOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions& options,
std::string* error_out) {
return CreateSdp(
[this, options](CreateSessionDescriptionObserver* observer) {
pc()->CreateOffer(observer, options);
},
error_out);
}
std::unique_ptr<SessionDescriptionInterface>
PeerConnectionWrapper::CreateOfferAndSetAsLocal() {
return CreateOfferAndSetAsLocal(RTCOfferAnswerOptions());
}
std::unique_ptr<SessionDescriptionInterface>
PeerConnectionWrapper::CreateOfferAndSetAsLocal(
const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
auto offer = CreateOffer(options);
if (!offer) {
return nullptr;
}
EXPECT_TRUE(SetLocalDescription(CloneSessionDescription(offer.get())));
return offer;
}
std::unique_ptr<SessionDescriptionInterface>
PeerConnectionWrapper::CreateAnswer() {
return CreateAnswer(RTCOfferAnswerOptions());
}
std::unique_ptr<SessionDescriptionInterface>
PeerConnectionWrapper::CreateAnswer(
const PeerConnectionInterface::RTCOfferAnswerOptions& options,
std::string* error_out) {
return CreateSdp(
[this, options](CreateSessionDescriptionObserver* observer) {
pc()->CreateAnswer(observer, options);
},
error_out);
}
std::unique_ptr<SessionDescriptionInterface>
PeerConnectionWrapper::CreateAnswerAndSetAsLocal() {
return CreateAnswerAndSetAsLocal(RTCOfferAnswerOptions());
}
std::unique_ptr<SessionDescriptionInterface>
PeerConnectionWrapper::CreateAnswerAndSetAsLocal(
const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
auto answer = CreateAnswer(options);
if (!answer) {
return nullptr;
}
EXPECT_TRUE(SetLocalDescription(CloneSessionDescription(answer.get())));
return answer;
}
std::unique_ptr<SessionDescriptionInterface> PeerConnectionWrapper::CreateSdp(
rtc::FunctionView<void(CreateSessionDescriptionObserver*)> fn,
std::string* error_out) {
rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer(
new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>());
fn(observer);
EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout);
if (error_out && !observer->result()) {
*error_out = observer->error();
}
return observer->MoveDescription();
}
bool PeerConnectionWrapper::SetLocalDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
std::string* error_out) {
return SetSdp(
[this, &desc](SetSessionDescriptionObserver* observer) {
pc()->SetLocalDescription(observer, desc.release());
},
error_out);
}
bool PeerConnectionWrapper::SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
std::string* error_out) {
return SetSdp(
[this, &desc](SetSessionDescriptionObserver* observer) {
pc()->SetRemoteDescription(observer, desc.release());
},
error_out);
}
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
bool PeerConnectionWrapper::SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
RTCError* error_out) {
rtc::scoped_refptr<MockSetRemoteDescriptionObserver> observer =
new MockSetRemoteDescriptionObserver();
pc()->SetRemoteDescription(std::move(desc), observer);
EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout);
bool ok = observer->error().ok();
if (error_out)
*error_out = std::move(observer->error());
return ok;
}
bool PeerConnectionWrapper::SetSdp(
rtc::FunctionView<void(SetSessionDescriptionObserver*)> fn,
std::string* error_out) {
rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
fn(observer);
EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout);
if (error_out && !observer->result()) {
*error_out = observer->error();
}
return observer->result();
}
bool PeerConnectionWrapper::ExchangeOfferAnswerWith(
PeerConnectionWrapper* answerer) {
return ExchangeOfferAnswerWith(answerer, RTCOfferAnswerOptions(),
RTCOfferAnswerOptions());
}
bool PeerConnectionWrapper::ExchangeOfferAnswerWith(
PeerConnectionWrapper* answerer,
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_options,
const PeerConnectionInterface::RTCOfferAnswerOptions& answer_options) {
RTC_DCHECK(answerer);
if (answerer == this) {
RTC_LOG(LS_ERROR) << "Cannot exchange offer/answer with ourself!";
return false;
}
auto offer = CreateOffer(offer_options);
EXPECT_TRUE(offer);
if (!offer) {
return false;
}
bool set_local_offer =
SetLocalDescription(CloneSessionDescription(offer.get()));
EXPECT_TRUE(set_local_offer);
if (!set_local_offer) {
return false;
}
bool set_remote_offer = answerer->SetRemoteDescription(std::move(offer));
EXPECT_TRUE(set_remote_offer);
if (!set_remote_offer) {
return false;
}
auto answer = answerer->CreateAnswer(answer_options);
EXPECT_TRUE(answer);
if (!answer) {
return false;
}
bool set_local_answer =
answerer->SetLocalDescription(CloneSessionDescription(answer.get()));
EXPECT_TRUE(set_local_answer);
if (!set_local_answer) {
return false;
}
bool set_remote_answer = SetRemoteDescription(std::move(answer));
EXPECT_TRUE(set_remote_answer);
return set_remote_answer;
}
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<RtpTransceiverInterface>
PeerConnectionWrapper::AddTransceiver(cricket::MediaType media_type) {
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> result =
pc()->AddTransceiver(media_type);
EXPECT_EQ(RTCErrorType::NONE, result.error().type());
return result.MoveValue();
}
rtc::scoped_refptr<RtpTransceiverInterface>
PeerConnectionWrapper::AddTransceiver(cricket::MediaType media_type,
const RtpTransceiverInit& init) {
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> result =
pc()->AddTransceiver(media_type, init);
EXPECT_EQ(RTCErrorType::NONE, result.error().type());
return result.MoveValue();
}
rtc::scoped_refptr<RtpTransceiverInterface>
PeerConnectionWrapper::AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> result =
pc()->AddTransceiver(track);
EXPECT_EQ(RTCErrorType::NONE, result.error().type());
return result.MoveValue();
}
rtc::scoped_refptr<RtpTransceiverInterface>
PeerConnectionWrapper::AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init) {
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> result =
pc()->AddTransceiver(track, init);
EXPECT_EQ(RTCErrorType::NONE, result.error().type());
return result.MoveValue();
}
rtc::scoped_refptr<AudioTrackInterface> PeerConnectionWrapper::CreateAudioTrack(
const std::string& label) {
return pc_factory()->CreateAudioTrack(label, nullptr);
}
rtc::scoped_refptr<VideoTrackInterface> PeerConnectionWrapper::CreateVideoTrack(
const std::string& label) {
return pc_factory()->CreateVideoTrack(label, FakeVideoTrackSource::Create());
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<RtpSenderInterface> PeerConnectionWrapper::AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids) {
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> result =
pc()->AddTrack(track, stream_ids);
EXPECT_EQ(RTCErrorType::NONE, result.error().type());
return result.MoveValue();
}
rtc::scoped_refptr<RtpSenderInterface> PeerConnectionWrapper::AddAudioTrack(
const std::string& track_label,
const std::vector<std::string>& stream_ids) {
return AddTrack(CreateAudioTrack(track_label), stream_ids);
}
rtc::scoped_refptr<RtpSenderInterface> PeerConnectionWrapper::AddVideoTrack(
const std::string& track_label,
const std::vector<std::string>& stream_ids) {
return AddTrack(CreateVideoTrack(track_label), stream_ids);
}
rtc::scoped_refptr<DataChannelInterface>
PeerConnectionWrapper::CreateDataChannel(const std::string& label) {
return pc()->CreateDataChannel(label, nullptr);
}
PeerConnectionInterface::SignalingState
PeerConnectionWrapper::signaling_state() {
return pc()->signaling_state();
}
bool PeerConnectionWrapper::IsIceGatheringDone() {
return observer()->ice_gathering_complete_;
}
bool PeerConnectionWrapper::IsIceConnected() {
return observer()->ice_connected_;
}
rtc::scoped_refptr<const webrtc::RTCStatsReport>
PeerConnectionWrapper::GetStats() {
rtc::scoped_refptr<webrtc::MockRTCStatsCollectorCallback> callback(
new rtc::RefCountedObject<webrtc::MockRTCStatsCollectorCallback>());
pc()->GetStats(callback);
EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout);
return callback->report();
}
} // namespace webrtc