Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 238 insertions, 0 deletions

diff --git a/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.mm b/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.mm
new file mode 100644
index 0000000000..90bcfcc35b
--- /dev/null
+++ b/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.mm
@@ -0,0 +1,238 @@
+/*
+ *  Copyright 2018 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 "examples/objcnativeapi/objc/objc_call_client.h"
+
+#include <memory>
+#include <utility>
+
+#import "sdk/objc/base/RTCVideoRenderer.h"
+#import "sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h"
+#import "sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h"
+#import "sdk/objc/helpers/RTCCameraPreviewView.h"
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/peer_connection_interface.h"
+#include "api/rtc_event_log/rtc_event_log_factory.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "media/engine/webrtc_media_engine.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "sdk/objc/native/api/video_capturer.h"
+#include "sdk/objc/native/api/video_decoder_factory.h"
+#include "sdk/objc/native/api/video_encoder_factory.h"
+#include "sdk/objc/native/api/video_renderer.h"
+
+namespace webrtc_examples {
+
+namespace {
+
+class CreateOfferObserver : public webrtc::CreateSessionDescriptionObserver {
+ public:
+  explicit CreateOfferObserver(rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc);
+
+  void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
+  void OnFailure(webrtc::RTCError error) override;
+
+ private:
+  const rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_;
+};
+
+class SetRemoteSessionDescriptionObserver : public webrtc::SetRemoteDescriptionObserverInterface {
+ public:
+  void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override;
+};
+
+class SetLocalSessionDescriptionObserver : public webrtc::SetLocalDescriptionObserverInterface {
+ public:
+  void OnSetLocalDescriptionComplete(webrtc::RTCError error) override;
+};
+
+}  // namespace
+
+ObjCCallClient::ObjCCallClient()
+    : call_started_(false), pc_observer_(std::make_unique<PCObserver>(this)) {
+  thread_checker_.Detach();
+  CreatePeerConnectionFactory();
+}
+
+void ObjCCallClient::Call(RTC_OBJC_TYPE(RTCVideoCapturer) * capturer,
+                          id<RTC_OBJC_TYPE(RTCVideoRenderer)> remote_renderer) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  webrtc::MutexLock lock(&pc_mutex_);
+  if (call_started_) {
+    RTC_LOG(LS_WARNING) << "Call already started.";
+    return;
+  }
+  call_started_ = true;
+
+  remote_sink_ = webrtc::ObjCToNativeVideoRenderer(remote_renderer);
+
+  video_source_ =
+      webrtc::ObjCToNativeVideoCapturer(capturer, signaling_thread_.get(), worker_thread_.get());
+
+  CreatePeerConnection();
+  Connect();
+}
+
+void ObjCCallClient::Hangup() {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  call_started_ = false;
+
+  {
+    webrtc::MutexLock lock(&pc_mutex_);
+    if (pc_ != nullptr) {
+      pc_->Close();
+      pc_ = nullptr;
+    }
+  }
+
+  remote_sink_ = nullptr;
+  video_source_ = nullptr;
+}
+
+void ObjCCallClient::CreatePeerConnectionFactory() {
+  network_thread_ = rtc::Thread::CreateWithSocketServer();
+  network_thread_->SetName("network_thread", nullptr);
+  RTC_CHECK(network_thread_->Start()) << "Failed to start thread";
+
+  worker_thread_ = rtc::Thread::Create();
+  worker_thread_->SetName("worker_thread", nullptr);
+  RTC_CHECK(worker_thread_->Start()) << "Failed to start thread";
+
+  signaling_thread_ = rtc::Thread::Create();
+  signaling_thread_->SetName("signaling_thread", nullptr);
+  RTC_CHECK(signaling_thread_->Start()) << "Failed to start thread";
+
+  webrtc::PeerConnectionFactoryDependencies dependencies;
+  dependencies.network_thread = network_thread_.get();
+  dependencies.worker_thread = worker_thread_.get();
+  dependencies.signaling_thread = signaling_thread_.get();
+  dependencies.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
+  cricket::MediaEngineDependencies media_deps;
+  media_deps.task_queue_factory = dependencies.task_queue_factory.get();
+  media_deps.audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
+  media_deps.audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
+  media_deps.video_encoder_factory = webrtc::ObjCToNativeVideoEncoderFactory(
+      [[RTC_OBJC_TYPE(RTCDefaultVideoEncoderFactory) alloc] init]);
+  media_deps.video_decoder_factory = webrtc::ObjCToNativeVideoDecoderFactory(
+      [[RTC_OBJC_TYPE(RTCDefaultVideoDecoderFactory) alloc] init]);
+  media_deps.audio_processing = webrtc::AudioProcessingBuilder().Create();
+  dependencies.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
+  RTC_LOG(LS_INFO) << "Media engine created: " << dependencies.media_engine.get();
+  dependencies.call_factory = webrtc::CreateCallFactory();
+  dependencies.event_log_factory =
+      std::make_unique<webrtc::RtcEventLogFactory>(dependencies.task_queue_factory.get());
+  pcf_ = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
+  RTC_LOG(LS_INFO) << "PeerConnectionFactory created: " << pcf_.get();
+}
+
+void ObjCCallClient::CreatePeerConnection() {
+  webrtc::MutexLock lock(&pc_mutex_);
+  webrtc::PeerConnectionInterface::RTCConfiguration config;
+  config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
+  // Encryption has to be disabled for loopback to work.
+  webrtc::PeerConnectionFactoryInterface::Options options;
+  options.disable_encryption = true;
+  pcf_->SetOptions(options);
+  webrtc::PeerConnectionDependencies pc_dependencies(pc_observer_.get());
+  pc_ = pcf_->CreatePeerConnectionOrError(config, std::move(pc_dependencies)).MoveValue();
+  RTC_LOG(LS_INFO) << "PeerConnection created: " << pc_.get();
+
+  rtc::scoped_refptr<webrtc::VideoTrackInterface> local_video_track =
+      pcf_->CreateVideoTrack(video_source_, "video");
+  pc_->AddTransceiver(local_video_track);
+  RTC_LOG(LS_INFO) << "Local video sink set up: " << local_video_track.get();
+
+  for (const rtc::scoped_refptr<webrtc::RtpTransceiverInterface>& tranceiver :
+       pc_->GetTransceivers()) {
+    rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track = tranceiver->receiver()->track();
+    if (track && track->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) {
+      static_cast<webrtc::VideoTrackInterface*>(track.get())
+          ->AddOrUpdateSink(remote_sink_.get(), rtc::VideoSinkWants());
+      RTC_LOG(LS_INFO) << "Remote video sink set up: " << track.get();
+      break;
+    }
+  }
+}
+
+void ObjCCallClient::Connect() {
+  webrtc::MutexLock lock(&pc_mutex_);
+  pc_->CreateOffer(rtc::make_ref_counted<CreateOfferObserver>(pc_).get(),
+                   webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
+}
+
+ObjCCallClient::PCObserver::PCObserver(ObjCCallClient* client) : client_(client) {}
+
+void ObjCCallClient::PCObserver::OnSignalingChange(
+    webrtc::PeerConnectionInterface::SignalingState new_state) {
+  RTC_LOG(LS_INFO) << "OnSignalingChange: " << new_state;
+}
+
+void ObjCCallClient::PCObserver::OnDataChannel(
+    rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) {
+  RTC_LOG(LS_INFO) << "OnDataChannel";
+}
+
+void ObjCCallClient::PCObserver::OnRenegotiationNeeded() {
+  RTC_LOG(LS_INFO) << "OnRenegotiationNeeded";
+}
+
+void ObjCCallClient::PCObserver::OnIceConnectionChange(
+    webrtc::PeerConnectionInterface::IceConnectionState new_state) {
+  RTC_LOG(LS_INFO) << "OnIceConnectionChange: " << new_state;
+}
+
+void ObjCCallClient::PCObserver::OnIceGatheringChange(
+    webrtc::PeerConnectionInterface::IceGatheringState new_state) {
+  RTC_LOG(LS_INFO) << "OnIceGatheringChange: " << new_state;
+}
+
+void ObjCCallClient::PCObserver::OnIceCandidate(const webrtc::IceCandidateInterface* candidate) {
+  RTC_LOG(LS_INFO) << "OnIceCandidate: " << candidate->server_url();
+  webrtc::MutexLock lock(&client_->pc_mutex_);
+  RTC_DCHECK(client_->pc_ != nullptr);
+  client_->pc_->AddIceCandidate(candidate);
+}
+
+CreateOfferObserver::CreateOfferObserver(rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc)
+    : pc_(pc) {}
+
+void CreateOfferObserver::OnSuccess(webrtc::SessionDescriptionInterface* desc) {
+  std::string sdp;
+  desc->ToString(&sdp);
+  RTC_LOG(LS_INFO) << "Created offer: " << sdp;
+
+  // Ownership of desc was transferred to us, now we transfer it forward.
+  pc_->SetLocalDescription(absl::WrapUnique(desc),
+                           rtc::make_ref_counted<SetLocalSessionDescriptionObserver>());
+
+  // Generate a fake answer.
+  std::unique_ptr<webrtc::SessionDescriptionInterface> answer(
+      webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp));
+  pc_->SetRemoteDescription(std::move(answer),
+                            rtc::make_ref_counted<SetRemoteSessionDescriptionObserver>());
+}
+
+void CreateOfferObserver::OnFailure(webrtc::RTCError error) {
+  RTC_LOG(LS_INFO) << "Failed to create offer: " << error.message();
+}
+
+void SetRemoteSessionDescriptionObserver::OnSetRemoteDescriptionComplete(webrtc::RTCError error) {
+  RTC_LOG(LS_INFO) << "Set remote description: " << error.message();
+}
+
+void SetLocalSessionDescriptionObserver::OnSetLocalDescriptionComplete(webrtc::RTCError error) {
+  RTC_LOG(LS_INFO) << "Set local description: " << error.message();
+}
+
+}  // namespace webrtc_examples