/* * 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 SignalSamplesRead; protected: SoundInputStreamInterface(); private: RTC_DISALLOW_COPY_AND_ASSIGN(SoundInputStreamInterface); }; } // namespace rtc #endif // WEBRTC_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_