3c089d751e
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition. * DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN * DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN * DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS Related CL: https://codereview.webrtc.org/1335923002/ BUG=chromium:468375 NOTRY=true Review URL: https://codereview.webrtc.org/1345433002 Cr-Commit-Position: refs/heads/master@{#9953}
104 lines
2.9 KiB
C++
104 lines
2.9 KiB
C++
/*
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* Copyright 2004 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_SOUND_ALSASOUNDSYSTEM_H_
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#define WEBRTC_SOUND_ALSASOUNDSYSTEM_H_
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#include "webrtc/sound/alsasymboltable.h"
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#include "webrtc/sound/soundsysteminterface.h"
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#include "webrtc/base/constructormagic.h"
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namespace rtc {
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class AlsaStream;
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class AlsaInputStream;
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class AlsaOutputStream;
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// Sound system implementation for ALSA, the predominant sound device API on
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// Linux (but typically not used directly by applications anymore).
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class AlsaSoundSystem : public SoundSystemInterface {
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friend class AlsaStream;
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friend class AlsaInputStream;
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friend class AlsaOutputStream;
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public:
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static SoundSystemInterface *Create() {
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return new AlsaSoundSystem();
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}
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AlsaSoundSystem();
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virtual ~AlsaSoundSystem();
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virtual bool Init();
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virtual void Terminate();
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virtual bool EnumeratePlaybackDevices(SoundDeviceLocatorList *devices);
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virtual bool EnumerateCaptureDevices(SoundDeviceLocatorList *devices);
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virtual bool GetDefaultPlaybackDevice(SoundDeviceLocator **device);
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virtual bool GetDefaultCaptureDevice(SoundDeviceLocator **device);
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virtual SoundOutputStreamInterface *OpenPlaybackDevice(
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const SoundDeviceLocator *device,
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const OpenParams ¶ms);
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virtual SoundInputStreamInterface *OpenCaptureDevice(
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const SoundDeviceLocator *device,
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const OpenParams ¶ms);
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virtual const char *GetName() const;
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private:
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bool IsInitialized() { return initialized_; }
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bool EnumerateDevices(SoundDeviceLocatorList *devices,
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bool capture_not_playback);
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bool GetDefaultDevice(SoundDeviceLocator **device);
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static size_t FrameSize(const OpenParams ¶ms);
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template <typename StreamInterface>
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StreamInterface *OpenDevice(
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const SoundDeviceLocator *device,
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const OpenParams ¶ms,
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snd_pcm_stream_t type,
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StreamInterface *(AlsaSoundSystem::*start_fn)(
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snd_pcm_t *handle,
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size_t frame_size,
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int wait_timeout_ms,
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int flags,
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int freq));
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SoundOutputStreamInterface *StartOutputStream(
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snd_pcm_t *handle,
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size_t frame_size,
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int wait_timeout_ms,
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int flags,
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int freq);
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SoundInputStreamInterface *StartInputStream(
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snd_pcm_t *handle,
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size_t frame_size,
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int wait_timeout_ms,
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int flags,
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int freq);
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const char *GetError(int err);
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bool initialized_;
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AlsaSymbolTable symbol_table_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AlsaSoundSystem);
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};
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} // namespace rtc
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#endif // WEBRTC_SOUND_ALSASOUNDSYSTEM_H_
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