| Index: webrtc/sound/soundoutputstreaminterface.h
|
| diff --git a/webrtc/sound/soundoutputstreaminterface.h b/webrtc/sound/soundoutputstreaminterface.h
|
| deleted file mode 100644
|
| index a94147b838469cdce2ec239896daf25cbcfdc4e4..0000000000000000000000000000000000000000
|
| --- a/webrtc/sound/soundoutputstreaminterface.h
|
| +++ /dev/null
|
| @@ -1,72 +0,0 @@
|
| -/*
|
| - * 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_SOUNDOUTPUTSTREAMINTERFACE_H_
|
| -#define WEBRTC_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_
|
| -
|
| -#include "webrtc/base/constructormagic.h"
|
| -#include "webrtc/base/sigslot.h"
|
| -
|
| -namespace rtc {
|
| -
|
| -// Interface for outputting a stream to a playback device.
|
| -// Semantics and thread-safety of EnableBufferMonitoring()/
|
| -// DisableBufferMonitoring() are the same as for rtc::Worker.
|
| -class SoundOutputStreamInterface {
|
| - public:
|
| - virtual ~SoundOutputStreamInterface();
|
| -
|
| - // Enables monitoring the available buffer space on the current thread.
|
| - virtual bool EnableBufferMonitoring() = 0;
|
| - // Disables the monitoring.
|
| - virtual bool DisableBufferMonitoring() = 0;
|
| -
|
| - // Write the given samples to the devices. If currently monitoring then this
|
| - // may only be called from the monitoring thread.
|
| - virtual bool WriteSamples(const void *sample_data,
|
| - size_t size) = 0;
|
| -
|
| - // Retrieves the current output 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 output 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 monitoring then this may only be
|
| - // called from the monitoring thread.
|
| - virtual bool Close() = 0;
|
| -
|
| - // Get the latency of the stream.
|
| - virtual int LatencyUsecs() = 0;
|
| -
|
| - // Notifies the producer of the available buffer space for writes.
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| - // It fires continuously as long as the space is greater than zero.
|
| - // The first parameter is the amount of buffer space available for data to
|
| - // be written (i.e., the maximum amount of data that can be written right now
|
| - // with WriteSamples() without blocking).
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| - // The 2nd parameter is the stream that is issuing the callback.
|
| - sigslot::signal2<size_t, SoundOutputStreamInterface *> SignalBufferSpace;
|
| -
|
| - protected:
|
| - SoundOutputStreamInterface();
|
| -
|
| - private:
|
| - RTC_DISALLOW_COPY_AND_ASSIGN(SoundOutputStreamInterface);
|
| -};
|
| -
|
| -} // namespace rtc
|
| -
|
| -#endif // WEBRTC_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_
|
|
|