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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_