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