webrtc/examples/objcnativeapi/objc/objccallclient.mm
Jiawei Ou c2ebe21ba9 Reland "Use the factory instead of using the builtin code path in VideoCodecInitializer"
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782

This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.

Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.

One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.

Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
2018-11-08 19:10:47 +00:00

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9.3 KiB
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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/objccallclient.h"
#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 "absl/memory/memory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/peerconnectioninterface.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "media/engine/webrtcmediaengine.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_(absl::make_unique<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<webrtc::VideoBitrateAllocatorFactory> videoBitrateAllocatorFactory =
webrtc::CreateBuiltinVideoBitrateAllocatorFactory();
std::unique_ptr<cricket::MediaEngineInterface> media_engine =
cricket::WebRtcMediaEngineFactory::Create(nullptr /* adm */,
webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(),
std::move(videoEncoderFactory),
std::move(videoDecoderFactory),
std::move(videoBitrateAllocatorFactory),
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(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