Files
platform-external-webrtc/examples/objcnativeapi/objc/objc_call_client.mm
Mirko Bonadei a81e9c82fc Wrap WebRTC OBJC API types with RTC_OBJC_TYPE.
This CL introduced 2 new macros that affect the WebRTC OBJC API symbols:

- RTC_OBJC_TYPE_PREFIX:
  Macro used to prepend a prefix to the API types that are exported with
  RTC_OBJC_EXPORT.

  Clients can patch the definition of this macro locally and build
  WebRTC.framework with their own prefix in case symbol clashing is a
  problem.

  This macro must only be defined by changing the value in
  sdk/objc/base/RTCMacros.h  and not on via compiler flag to ensure
  it has a unique value.

- RCT_OBJC_TYPE:
  Macro used internally to reference API types. Declaring an API type
  without using this macro will not include the declared type in the
  set of types that will be affected by the configurable
  RTC_OBJC_TYPE_PREFIX.

Manual changes:
https://webrtc-review.googlesource.com/c/src/+/173781/5..10

The auto-generated changes in PS#5 have been done with:
https://webrtc-review.googlesource.com/c/src/+/174061.

Bug: None
Change-Id: I0d54ca94db764fb3b6cb4365873f79e14cd879b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31153}
2020-05-04 15:01:26 +00:00

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/*
* 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::SetSessionDescriptionObserver {
public:
void OnSuccess() override;
void OnFailure(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_);
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";
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_;
}
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;
webrtc::PeerConnectionDependencies pc_dependencies(pc_observer_.get());
pc_ = pcf_->CreatePeerConnection(config, std::move(pc_dependencies));
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(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::OnSuccess() {
RTC_LOG(LS_INFO) << "Set local description success!";
}
void SetLocalSessionDescriptionObserver::OnFailure(webrtc::RTCError error) {
RTC_LOG(LS_INFO) << "Set local description failure: " << error.message();
}
} // namespace webrtc_examples