66a3582170
BUG=3379 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22379004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6986 4adac7df-926f-26a2-2b94-8c16560cd09d
73 lines
2.7 KiB
C++
73 lines
2.7 KiB
C++
/*
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* Copyright 2004 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_
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#define WEBRTC_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/sigslot.h"
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namespace rtc {
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// Interface for outputting a stream to a playback device.
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// Semantics and thread-safety of EnableBufferMonitoring()/
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// DisableBufferMonitoring() are the same as for rtc::Worker.
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class SoundOutputStreamInterface {
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public:
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virtual ~SoundOutputStreamInterface() {}
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// Enables monitoring the available buffer space on the current thread.
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virtual bool EnableBufferMonitoring() = 0;
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// Disables the monitoring.
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virtual bool DisableBufferMonitoring() = 0;
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// Write the given samples to the devices. If currently monitoring then this
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// may only be called from the monitoring thread.
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virtual bool WriteSamples(const void *sample_data,
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size_t size) = 0;
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// Retrieves the current output volume for this stream. Nominal range is
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// defined by SoundSystemInterface::k(Max|Min)Volume, but values exceeding the
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// max may be possible in some implementations. This call retrieves the actual
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// volume currently in use by the OS, not a cached value from a previous
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// (Get|Set)Volume() call.
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virtual bool GetVolume(int *volume) = 0;
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// Changes the output volume for this stream. Nominal range is defined by
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// SoundSystemInterface::k(Max|Min)Volume. The effect of exceeding kMaxVolume
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// is implementation-defined.
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virtual bool SetVolume(int volume) = 0;
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// Closes this stream object. If currently monitoring then this may only be
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// called from the monitoring thread.
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virtual bool Close() = 0;
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// Get the latency of the stream.
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virtual int LatencyUsecs() = 0;
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// Notifies the producer of the available buffer space for writes.
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// It fires continuously as long as the space is greater than zero.
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// The first parameter is the amount of buffer space available for data to
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// be written (i.e., the maximum amount of data that can be written right now
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// with WriteSamples() without blocking).
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// The 2nd parameter is the stream that is issuing the callback.
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sigslot::signal2<size_t, SoundOutputStreamInterface *> SignalBufferSpace;
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protected:
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SoundOutputStreamInterface() {}
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private:
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DISALLOW_COPY_AND_ASSIGN(SoundOutputStreamInterface);
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};
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} // namespace rtc
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#endif // WEBRTC_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_
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