1/*
2 * Copyright (C) 2017-2019 Apple Inc.
3 *
4 * Redistribution and use in source and binary forms, with or without
5 * modification, are permitted, provided that the following conditions
6 * are required to be met:
7 *
8 * 1. Redistributions of source code must retain the above copyright
9 * notice, this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright
11 * notice, this list of conditions and the following disclaimer in the
12 * documentation and/or other materials provided with the distribution.
13 * 3. Neither the name of Apple Inc. nor the names of
14 * its contributors may be used to endorse or promote products derived
15 * from this software without specific prior written permission.
16 *
17 * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS "AS IS" AND ANY
18 * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
19 * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
20 * DISCLAIMED. IN NO EVENT SHALL APPLE INC. AND ITS CONTRIBUTORS BE LIABLE FOR
21 * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
22 * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
23 * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
24 * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
25 * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
26 * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
27 */
28
29#pragma once
30
31
32#if USE(LIBWEBRTC)
33
34#include "LibWebRTCMacros.h"
35#include "MediaStreamTrackPrivate.h"
36#include "Timer.h"
37
38ALLOW_UNUSED_PARAMETERS_BEGIN
39
40#include <webrtc/api/mediastreaminterface.h>
41
42ALLOW_UNUSED_PARAMETERS_END
43
44#include <wtf/LoggerHelper.h>
45#include <wtf/ThreadSafeRefCounted.h>
46
47namespace webrtc {
48class AudioTrackInterface;
49class AudioTrackSinkInterface;
50}
51
52namespace WebCore {
53
54class RealtimeOutgoingAudioSource
55 : public ThreadSafeRefCounted<RealtimeOutgoingAudioSource, WTF::DestructionThread::Main>
56 , public webrtc::AudioSourceInterface
57 , private MediaStreamTrackPrivate::Observer
58#if !RELEASE_LOG_DISABLED
59 , private LoggerHelper
60#endif
61{
62public:
63 static Ref<RealtimeOutgoingAudioSource> create(Ref<MediaStreamTrackPrivate>&& audioSource);
64
65 ~RealtimeOutgoingAudioSource();
66
67 void stop() { unobserveSource(); }
68
69 bool setSource(Ref<MediaStreamTrackPrivate>&&);
70 MediaStreamTrackPrivate& source() const { return m_audioSource.get(); }
71
72protected:
73 explicit RealtimeOutgoingAudioSource(Ref<MediaStreamTrackPrivate>&&);
74
75 void unobserveSource();
76
77 virtual void pullAudioData() { }
78
79 bool isSilenced() const { return m_muted || !m_enabled; }
80
81 void sendAudioFrames(const void* audioData, int bitsPerSample, int sampleRate, size_t numberOfChannels, size_t numberOfFrames);
82
83#if !RELEASE_LOG_DISABLED
84 // LoggerHelper API
85 const Logger& logger() const final { return m_logger.get(); }
86 const void* logIdentifier() const final { return m_logIdentifier; }
87 const char* logClassName() const final { return "RealtimeOutgoingAudioSource"; }
88 WTFLogChannel& logChannel() const final;
89#endif
90
91private:
92 // webrtc::AudioSourceInterface API
93 void AddSink(webrtc::AudioTrackSinkInterface*) final;
94 void RemoveSink(webrtc::AudioTrackSinkInterface*) final;
95
96 void AddRef() const final { ref(); }
97 rtc::RefCountReleaseStatus Release() const final
98 {
99 auto result = refCount() - 1;
100 deref();
101 return result ? rtc::RefCountReleaseStatus::kOtherRefsRemained : rtc::RefCountReleaseStatus::kDroppedLastRef;
102 }
103
104 SourceState state() const final { return kLive; }
105 bool remote() const final { return false; }
106 void RegisterObserver(webrtc::ObserverInterface*) final { }
107 void UnregisterObserver(webrtc::ObserverInterface*) final { }
108
109 void observeSource();
110
111 void sourceMutedChanged();
112 void sourceEnabledChanged();
113 virtual void audioSamplesAvailable(const MediaTime&, const PlatformAudioData&, const AudioStreamDescription&, size_t) { };
114
115 virtual bool isReachingBufferedAudioDataHighLimit() { return false; };
116 virtual bool isReachingBufferedAudioDataLowLimit() { return false; };
117 virtual bool hasBufferedEnoughData() { return false; };
118
119 // MediaStreamTrackPrivate::Observer API
120 void trackMutedChanged(MediaStreamTrackPrivate&) final { sourceMutedChanged(); }
121 void trackEnabledChanged(MediaStreamTrackPrivate&) final { sourceEnabledChanged(); }
122 void audioSamplesAvailable(MediaStreamTrackPrivate&, const MediaTime& mediaTime, const PlatformAudioData& data, const AudioStreamDescription& description, size_t sampleCount) { audioSamplesAvailable(mediaTime, data, description, sampleCount); }
123 void trackEnded(MediaStreamTrackPrivate&) final { }
124 void trackSettingsChanged(MediaStreamTrackPrivate&) final { }
125
126 void initializeConverter();
127
128 Ref<MediaStreamTrackPrivate> m_audioSource;
129 bool m_muted { false };
130 bool m_enabled { true };
131
132 mutable RecursiveLock m_sinksLock;
133 HashSet<webrtc::AudioTrackSinkInterface*> m_sinks;
134
135#if !RELEASE_LOG_DISABLED
136 mutable Ref<const Logger> m_logger;
137 const void* m_logIdentifier;
138 size_t m_chunksSent { 0 };
139#endif
140};
141
142} // namespace WebCore
143
144#endif // USE(LIBWEBRTC)
145