| /* |
| * 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_ |