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
|
// Copyright 2016 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "remoting/protocol/webrtc_audio_stream.h"
#include "base/check.h"
#include "base/location.h"
#include "base/task/single_thread_task_runner.h"
#include "remoting/base/constants.h"
#include "remoting/protocol/audio_source.h"
#include "remoting/protocol/webrtc_audio_source_adapter.h"
#include "remoting/protocol/webrtc_transport.h"
#include "third_party/webrtc/api/media_stream_interface.h"
#include "third_party/webrtc/api/peer_connection_interface.h"
#include "third_party/webrtc/rtc_base/ref_count.h"
namespace remoting::protocol {
const char kAudioStreamLabel[] = "audio_stream";
const char kAudioTrackLabel[] = "system_audio";
WebrtcAudioStream::WebrtcAudioStream() = default;
WebrtcAudioStream::~WebrtcAudioStream() = default;
void WebrtcAudioStream::Start(
scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner,
std::unique_ptr<AudioSource> audio_source,
WebrtcTransport* webrtc_transport) {
DCHECK(webrtc_transport);
source_adapter_ =
new webrtc::RefCountedObject<WebrtcAudioSourceAdapter>(audio_task_runner);
source_adapter_->Start(std::move(audio_source));
scoped_refptr<webrtc::PeerConnectionFactoryInterface> peer_connection_factory(
webrtc_transport->peer_connection_factory());
peer_connection_ = webrtc_transport->peer_connection();
webrtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track =
peer_connection_factory->CreateAudioTrack(kAudioTrackLabel,
source_adapter_.get());
webrtc::RtpTransceiverInit init;
init.stream_ids = {kAudioStreamLabel};
// value() DCHECKs if AddTransceiver() fails, which only happens if a track
// was already added with the stream label.
auto transceiver =
peer_connection_->AddTransceiver(audio_track, init).value();
webrtc_transport->OnAudioTransceiverCreated(transceiver);
}
void WebrtcAudioStream::Pause(bool pause) {
source_adapter_->Pause(pause);
}
} // namespace remoting::protocol
|
