Support arbitrary video resolution in getUserMedia API
[WebKit-https.git] / Source / WebCore / platform / mediastream / RealtimeIncomingAudioSource.cpp
1 /*
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30
31 #include "config.h"
32 #include "RealtimeIncomingAudioSource.h"
33
34 #if USE(LIBWEBRTC)
35
36 #include "LibWebRTCAudioFormat.h"
37
38 namespace WebCore {
39
40 RealtimeIncomingAudioSource::RealtimeIncomingAudioSource(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
41     : RealtimeMediaSource(WTFMove(audioTrackId), RealtimeMediaSource::Type::Audio, String())
42     , m_audioTrack(WTFMove(audioTrack))
43 {
44     setName("remote audio");
45     notifyMutedChange(!m_audioTrack);
46 }
47
48 RealtimeIncomingAudioSource::~RealtimeIncomingAudioSource()
49 {
50     stop();
51 }
52
53 void RealtimeIncomingAudioSource::startProducingData()
54 {
55     if (m_audioTrack)
56         m_audioTrack->AddSink(this);
57 }
58
59 void RealtimeIncomingAudioSource::stopProducingData()
60 {
61     if (m_audioTrack)
62         m_audioTrack->RemoveSink(this);
63 }
64
65 void RealtimeIncomingAudioSource::setSourceTrack(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& track)
66 {
67     ASSERT(!m_audioTrack);
68     ASSERT(track);
69
70     m_audioTrack = WTFMove(track);
71     notifyMutedChange(!m_audioTrack);
72     if (isProducingData())
73         m_audioTrack->AddSink(this);
74 }
75
76 const RealtimeMediaSourceCapabilities& RealtimeIncomingAudioSource::capabilities()
77 {
78     return RealtimeMediaSourceCapabilities::emptyCapabilities();
79 }
80
81 const RealtimeMediaSourceSettings& RealtimeIncomingAudioSource::settings()
82 {
83     return m_currentSettings;
84 }
85
86 }
87
88 #endif // USE(LIBWEBRTC)