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1 /* | |
2 * Copyright 2004 The WebRTC Project Authors. All rights reserved. | |
3 * | |
4 * Use of this source code is governed by a BSD-style license | |
5 * that can be found in the LICENSE file in the root of the source | |
6 * tree. An additional intellectual property rights grant can be found | |
7 * in the file PATENTS. All contributing project authors may | |
8 * be found in the AUTHORS file in the root of the source tree. | |
9 */ | |
10 | |
11 #ifndef WEBRTC_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_ | |
12 #define WEBRTC_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_ | |
13 | |
14 #include "webrtc/base/constructormagic.h" | |
15 #include "webrtc/base/sigslot.h" | |
16 | |
17 namespace rtc { | |
18 | |
19 // Interface for outputting a stream to a playback device. | |
20 // Semantics and thread-safety of EnableBufferMonitoring()/ | |
21 // DisableBufferMonitoring() are the same as for rtc::Worker. | |
22 class SoundOutputStreamInterface { | |
23 public: | |
24 virtual ~SoundOutputStreamInterface(); | |
25 | |
26 // Enables monitoring the available buffer space on the current thread. | |
27 virtual bool EnableBufferMonitoring() = 0; | |
28 // Disables the monitoring. | |
29 virtual bool DisableBufferMonitoring() = 0; | |
30 | |
31 // Write the given samples to the devices. If currently monitoring then this | |
32 // may only be called from the monitoring thread. | |
33 virtual bool WriteSamples(const void *sample_data, | |
34 size_t size) = 0; | |
35 | |
36 // Retrieves the current output volume for this stream. Nominal range is | |
37 // defined by SoundSystemInterface::k(Max|Min)Volume, but values exceeding the | |
38 // max may be possible in some implementations. This call retrieves the actual | |
39 // volume currently in use by the OS, not a cached value from a previous | |
40 // (Get|Set)Volume() call. | |
41 virtual bool GetVolume(int *volume) = 0; | |
42 | |
43 // Changes the output volume for this stream. Nominal range is defined by | |
44 // SoundSystemInterface::k(Max|Min)Volume. The effect of exceeding kMaxVolume | |
45 // is implementation-defined. | |
46 virtual bool SetVolume(int volume) = 0; | |
47 | |
48 // Closes this stream object. If currently monitoring then this may only be | |
49 // called from the monitoring thread. | |
50 virtual bool Close() = 0; | |
51 | |
52 // Get the latency of the stream. | |
53 virtual int LatencyUsecs() = 0; | |
54 | |
55 // Notifies the producer of the available buffer space for writes. | |
56 // It fires continuously as long as the space is greater than zero. | |
57 // The first parameter is the amount of buffer space available for data to | |
58 // be written (i.e., the maximum amount of data that can be written right now | |
59 // with WriteSamples() without blocking). | |
60 // The 2nd parameter is the stream that is issuing the callback. | |
61 sigslot::signal2<size_t, SoundOutputStreamInterface *> SignalBufferSpace; | |
62 | |
63 protected: | |
64 SoundOutputStreamInterface(); | |
65 | |
66 private: | |
67 RTC_DISALLOW_COPY_AND_ASSIGN(SoundOutputStreamInterface); | |
68 }; | |
69 | |
70 } // namespace rtc | |
71 | |
72 #endif // WEBRTC_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_ | |
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