Add an example app for iOS native API.
Demonstrates how to use the iOS native API to wrap components into
C++ classes.
This CL also introduces a native API wrapper for the capturer.
The C++ code is forked from the corresponding CL for Android at
https://webrtc-review.googlesource.com/c/src/+/60540
Bug: webrtc:8832
Change-Id: I12d9f30e701c0222628e329218f6d5bfca26e6e0
Reviewed-on: https://webrtc-review.googlesource.com/61422
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22484}diff --git a/examples/objcnativeapi/objc/objccallclient.mm b/examples/objcnativeapi/objc/objccallclient.mm
new file mode 100644
index 0000000..68c58e2
--- /dev/null
+++ b/examples/objcnativeapi/objc/objccallclient.mm
@@ -0,0 +1,237 @@
+/*
+ * 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/objccallclient.h"
+
+#include <utility>
+
+#import <WebRTC/RTCCameraPreviewView.h>
+#import <WebRTC/RTCVideoCodecFactory.h>
+#import <WebRTC/RTCVideoRenderer.h>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/peerconnectioninterface.h"
+#include "media/engine/webrtcmediaengine.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/ptr_util.h"
+#include "sdk/objc/Framework/Native/api/video_capturer.h"
+#include "sdk/objc/Framework/Native/api/video_decoder_factory.h"
+#include "sdk/objc/Framework/Native/api/video_encoder_factory.h"
+#include "sdk/objc/Framework/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(const std::string& 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::SetSessionDescriptionObserver {
+ public:
+ void OnSuccess() override;
+ void OnFailure(const std::string& error) override;
+};
+
+} // namespace
+
+ObjCCallClient::ObjCCallClient()
+ : call_started_(false), pc_observer_(rtc::MakeUnique<PCObserver>(this)) {
+ thread_checker_.DetachFromThread();
+ CreatePeerConnectionFactory();
+}
+
+void ObjCCallClient::Call(RTCVideoCapturer* capturer, id<RTCVideoRenderer> remote_renderer) {
+ RTC_DCHECK_RUN_ON(&thread_checker_);
+
+ rtc::CritScope 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;
+
+ {
+ rtc::CritScope 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";
+
+ std::unique_ptr<webrtc::VideoDecoderFactory> videoDecoderFactory =
+ webrtc::ObjCToNativeVideoDecoderFactory([[RTCDefaultVideoDecoderFactory alloc] init]);
+ std::unique_ptr<webrtc::VideoEncoderFactory> videoEncoderFactory =
+ webrtc::ObjCToNativeVideoEncoderFactory([[RTCDefaultVideoEncoderFactory alloc] init]);
+
+ std::unique_ptr<cricket::MediaEngineInterface> media_engine =
+ cricket::WebRtcMediaEngineFactory::Create(nullptr /* adm */,
+ webrtc::CreateBuiltinAudioEncoderFactory(),
+ webrtc::CreateBuiltinAudioDecoderFactory(),
+ std::move(videoEncoderFactory),
+ std::move(videoDecoderFactory),
+ nullptr /* audio_mixer */,
+ webrtc::AudioProcessingBuilder().Create());
+ RTC_LOG(LS_INFO) << "Media engine created: " << media_engine.get();
+
+ pcf_ = webrtc::CreateModularPeerConnectionFactory(network_thread_.get(),
+ worker_thread_.get(),
+ signaling_thread_.get(),
+ std::move(media_engine),
+ webrtc::CreateCallFactory(),
+ webrtc::CreateRtcEventLogFactory());
+ RTC_LOG(LS_INFO) << "PeerConnectionFactory created: " << pcf_;
+}
+
+void ObjCCallClient::CreatePeerConnection() {
+ rtc::CritScope lock(&pc_mutex_);
+ webrtc::PeerConnectionInterface::RTCConfiguration config;
+ config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
+ // DTLS SRTP has to be disabled for loopback to work.
+ config.enable_dtls_srtp = false;
+ pc_ = pcf_->CreatePeerConnection(
+ config, nullptr /* port_allocator */, nullptr /* cert_generator */, pc_observer_.get());
+ RTC_LOG(LS_INFO) << "PeerConnection created: " << pc_;
+
+ rtc::scoped_refptr<webrtc::VideoTrackInterface> local_video_track =
+ pcf_->CreateVideoTrack("video", video_source_);
+ pc_->AddTransceiver(local_video_track);
+ RTC_LOG(LS_INFO) << "Local video sink set up: " << local_video_track;
+
+ 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;
+ break;
+ }
+ }
+}
+
+void ObjCCallClient::Connect() {
+ rtc::CritScope lock(&pc_mutex_);
+ pc_->CreateOffer(new rtc::RefCountedObject<CreateOfferObserver>(pc_),
+ 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();
+ rtc::CritScope 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(new rtc::RefCountedObject<SetLocalSessionDescriptionObserver>(), desc);
+
+ // Generate a fake answer.
+ std::unique_ptr<webrtc::SessionDescriptionInterface> answer(
+ webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp));
+ pc_->SetRemoteDescription(std::move(answer),
+ new rtc::RefCountedObject<SetRemoteSessionDescriptionObserver>());
+}
+
+void CreateOfferObserver::OnFailure(const std::string& error) {
+ RTC_LOG(LS_INFO) << "Failed to create offer: " << error;
+}
+
+void SetRemoteSessionDescriptionObserver::OnSetRemoteDescriptionComplete(webrtc::RTCError error) {
+ RTC_LOG(LS_INFO) << "Set remote description: " << error.message();
+}
+
+void SetLocalSessionDescriptionObserver::OnSuccess() {
+ RTC_LOG(LS_INFO) << "Set local description success!";
+}
+
+void SetLocalSessionDescriptionObserver::OnFailure(const std::string& error) {
+ RTC_LOG(LS_INFO) << "Set local description failure: " << error;
+}
+
+} // namespace webrtc_examples