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