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
|
// Copyright 2016 The Chromium Authors. All rights reserved.
// 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/location.h"
#include "base/logging.h"
#include "base/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/mediastreaminterface.h"
#include "third_party/webrtc/api/peerconnectioninterface.h"
#include "third_party/webrtc/base/refcount.h"
namespace remoting {
namespace protocol {
const char kAudioStreamLabel[] = "audio_stream";
const char kAudioTrackLabel[] = "system_audio";
WebrtcAudioStream::WebrtcAudioStream() {}
WebrtcAudioStream::~WebrtcAudioStream() {
if (stream_) {
for (const auto& track : stream_->GetAudioTracks()) {
stream_->RemoveTrack(track.get());
}
peer_connection_->RemoveStream(stream_.get());
}
}
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 rtc::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();
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track =
peer_connection_factory->CreateAudioTrack(kAudioTrackLabel,
source_adapter_.get());
stream_ = peer_connection_factory->CreateLocalMediaStream(kAudioStreamLabel);
// AddTrack() may fail only if there is another track with the same name,
// which is impossible because it's a brand new stream.
bool result = stream_->AddTrack(audio_track.get());
DCHECK(result);
// AddStream() may fail if there is another stream with the same name or when
// the PeerConnection is closed, neither is expected.
result = peer_connection_->AddStream(stream_.get());
DCHECK(result);
}
void WebrtcAudioStream::Pause(bool pause) {
source_adapter_->Pause(pause);
}
} // namespace protocol
} // namespace remoting
|
