/*
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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_SOUNDINPUTSTREAMINTERFACE_H_
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#define WEBRTC_SOUND_SOUNDINPUTSTREAMINTERFACE_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 consuming an input stream from a recording device.
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// Semantics and thread-safety of StartReading()/StopReading() are the same as
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// for rtc::Worker.
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class SoundInputStreamInterface {
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public:
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virtual ~SoundInputStreamInterface();
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// Starts the reading of samples on the current thread.
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virtual bool StartReading() = 0;
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// Stops the reading of samples.
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virtual bool StopReading() = 0;
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// Retrieves the current input 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 input 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 reading then this may only be
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// called from the reading 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 consumer of new data read from the device.
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// The first parameter is a pointer to the data read, and is only valid for
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// the duration of the call.
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// The second parameter is the amount of data read in bytes (i.e., the valid
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// length of the memory pointed to).
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// The 3rd parameter is the stream that is issuing the callback.
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sigslot::signal3<const void *, size_t,
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SoundInputStreamInterface *> SignalSamplesRead;
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protected:
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SoundInputStreamInterface();
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private:
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RTC_DISALLOW_COPY_AND_ASSIGN(SoundInputStreamInterface);
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};
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} // namespace rtc
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#endif // WEBRTC_SOUND_SOUNDINPUTSTREAMINTERFACE_H_
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