webrtc-sys 0.3.28

Unsafe bindings to libwebrtc
Documentation 
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143

/*
 * Copyright 2025 LiveKit, Inc.
 *
 * Licensed under the Apache License, Version 2.0 (the "License");
 * you may not use this file except in compliance with the License.
 * You may obtain a copy of the License at
 *
 *     http://www.apache.org/licenses/LICENSE-2.0
 *
 * Unless required by applicable law or agreed to in writing, software
 * distributed under the License is distributed on an "AS IS" BASIS,
 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
 * See the License for the specific language governing permissions and
 * limitations under the License.
 */

#include "livekit/peer_connection_factory.h"

#include <memory>
#include <utility>

#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/audio/builtin_audio_processing_builder.h"
#include "api/create_modular_peer_connection_factory.h"
#include "api/environment/environment_factory.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_error.h"
#include "api/enable_media.h"
#include "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "livekit/audio_device.h"
#include "livekit/audio_track.h"
#include "livekit/peer_connection.h"
#include "livekit/rtc_error.h"
#include "livekit/rtp_parameters.h"
#include "livekit/video_decoder_factory.h"
#include "livekit/video_encoder_factory.h"
#include "livekit/webrtc.h"
#include "rtc_base/thread.h"
#include "webrtc-sys/src/peer_connection.rs.h"
#include "webrtc-sys/src/peer_connection_factory.rs.h"

namespace livekit_ffi {

class PeerConnectionObserver;

PeerConnectionFactory::PeerConnectionFactory(
    std::shared_ptr<RtcRuntime> rtc_runtime)
    : rtc_runtime_(rtc_runtime),
    env_(webrtc::EnvironmentFactory().Create()) {
  RTC_LOG(LS_VERBOSE) << "PeerConnectionFactory::PeerConnectionFactory()";

  webrtc::PeerConnectionFactoryDependencies dependencies;
  dependencies.network_thread = rtc_runtime_->network_thread();
  dependencies.worker_thread = rtc_runtime_->worker_thread();
  dependencies.signaling_thread = rtc_runtime_->signaling_thread();
  dependencies.socket_factory = rtc_runtime_->network_thread()->socketserver();
  dependencies.event_log_factory = std::make_unique<webrtc::RtcEventLogFactory>();
  // TODO:
  // dependencies.trials = std::make_unique<webrtc::FieldTrialBasedConfig>();

  audio_device_ = rtc_runtime_->worker_thread()->BlockingCall([&] {
    return webrtc::make_ref_counted<livekit_ffi::AudioDevice>(env_);
  });

  dependencies.adm = audio_device_;

  dependencies.video_encoder_factory =
      std::move(std::make_unique<livekit_ffi::VideoEncoderFactory>());
  dependencies.video_decoder_factory =
      std::move(std::make_unique<livekit_ffi::VideoDecoderFactory>());
  dependencies.audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
  dependencies.audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
  dependencies.audio_processing_builder = std::make_unique<webrtc::BuiltinAudioProcessingBuilder>();

  webrtc::EnableMedia(dependencies);
  peer_factory_ =
      webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));

  if (peer_factory_.get() == nullptr) {
    RTC_LOG_ERR(LS_ERROR) << "Failed to create PeerConnectionFactory";
    return;
  }
}

PeerConnectionFactory::~PeerConnectionFactory() {
  RTC_LOG(LS_VERBOSE) << "PeerConnectionFactory::~PeerConnectionFactory()";

  peer_factory_ = nullptr;
  rtc_runtime_->worker_thread()->BlockingCall(
      [this] { audio_device_ = nullptr; });
}

std::shared_ptr<PeerConnection> PeerConnectionFactory::create_peer_connection(
    RtcConfiguration config,
    rust::Box<PeerConnectionObserverWrapper> observer) const {
  std::shared_ptr<PeerConnection> pc = std::make_shared<PeerConnection>(
      rtc_runtime_, peer_factory_, std::move(observer));

  if (!pc->Initialize(to_native_rtc_configuration(config))) {
    throw std::runtime_error(serialize_error(to_error(webrtc::RTCError(
        webrtc::RTCErrorType::INTERNAL_ERROR, "failed to initialize pc"))));
  }

  return pc;
}

std::shared_ptr<VideoTrack> PeerConnectionFactory::create_video_track(
    rust::String label,
    std::shared_ptr<VideoTrackSource> source) const {
  return std::static_pointer_cast<VideoTrack>(
      rtc_runtime_->get_or_create_media_stream_track(
          peer_factory_->CreateVideoTrack(source->get(), label.c_str())));
}

std::shared_ptr<AudioTrack> PeerConnectionFactory::create_audio_track(
    rust::String label,
    std::shared_ptr<AudioTrackSource> source) const {
  return std::static_pointer_cast<AudioTrack>(
      rtc_runtime_->get_or_create_media_stream_track(
          peer_factory_->CreateAudioTrack(label.c_str(), source->get().get())));
}

RtpCapabilities PeerConnectionFactory::rtp_sender_capabilities(
    MediaType type) const {
  return to_rust_rtp_capabilities(peer_factory_->GetRtpSenderCapabilities(
      static_cast<webrtc::MediaType>(type)));
}

RtpCapabilities PeerConnectionFactory::rtp_receiver_capabilities(
    MediaType type) const {
  return to_rust_rtp_capabilities(peer_factory_->GetRtpReceiverCapabilities(
      static_cast<webrtc::MediaType>(type)));
}

std::shared_ptr<PeerConnectionFactory> create_peer_connection_factory() {
  return std::make_shared<PeerConnectionFactory>(RtcRuntime::create());
}

}  // namespace livekit_ffi