
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}
178 lines
5.5 KiB
C++
178 lines
5.5 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_PULSEAUDIOSOUNDSYSTEM_H_
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#define WEBRTC_SOUND_PULSEAUDIOSOUNDSYSTEM_H_
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#ifdef HAVE_LIBPULSE
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#include "webrtc/sound/pulseaudiosymboltable.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 PulseAudioInputStream;
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class PulseAudioOutputStream;
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class PulseAudioStream;
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// Sound system implementation for PulseAudio, a cross-platform sound server
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// (but commonly used only on Linux, which is the only platform we support
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// it on).
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// Init(), Terminate(), and the destructor should never be invoked concurrently,
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// but all other methods are thread-safe.
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class PulseAudioSoundSystem : public SoundSystemInterface {
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friend class PulseAudioInputStream;
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friend class PulseAudioOutputStream;
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friend class PulseAudioStream;
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public:
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static SoundSystemInterface *Create() {
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return new PulseAudioSoundSystem();
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}
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PulseAudioSoundSystem();
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virtual ~PulseAudioSoundSystem();
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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();
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static void ConnectToPulseCallbackThunk(pa_context *context, void *userdata);
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void OnConnectToPulseCallback(pa_context *context, bool *connect_done);
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bool ConnectToPulse(pa_context *context);
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pa_context *CreateNewConnection();
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template <typename InfoStruct>
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bool EnumerateDevices(SoundDeviceLocatorList *devices,
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pa_operation *(*enumerate_fn)(
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pa_context *c,
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void (*callback_fn)(
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pa_context *c,
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const InfoStruct *i,
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int eol,
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void *userdata),
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void *userdata),
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void (*callback_fn)(
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pa_context *c,
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const InfoStruct *i,
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int eol,
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void *userdata));
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static void EnumeratePlaybackDevicesCallbackThunk(pa_context *unused,
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const pa_sink_info *info,
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int eol,
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void *userdata);
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static void EnumerateCaptureDevicesCallbackThunk(pa_context *unused,
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const pa_source_info *info,
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int eol,
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void *userdata);
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void OnEnumeratePlaybackDevicesCallback(
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SoundDeviceLocatorList *devices,
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const pa_sink_info *info,
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int eol);
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void OnEnumerateCaptureDevicesCallback(
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SoundDeviceLocatorList *devices,
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const pa_source_info *info,
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int eol);
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template <const char *(pa_server_info::*field)>
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static void GetDefaultDeviceCallbackThunk(
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pa_context *unused,
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const pa_server_info *info,
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void *userdata);
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template <const char *(pa_server_info::*field)>
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void OnGetDefaultDeviceCallback(
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const pa_server_info *info,
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SoundDeviceLocator **device);
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template <const char *(pa_server_info::*field)>
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bool GetDefaultDevice(SoundDeviceLocator **device);
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static void StreamStateChangedCallbackThunk(pa_stream *stream,
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void *userdata);
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void OnStreamStateChangedCallback(pa_stream *stream);
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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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const char *stream_name,
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StreamInterface *(PulseAudioSoundSystem::*connect_fn)(
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pa_stream *stream,
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const char *dev,
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int flags,
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pa_stream_flags_t pa_flags,
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int latency,
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const pa_sample_spec &spec));
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SoundOutputStreamInterface *ConnectOutputStream(
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pa_stream *stream,
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const char *dev,
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int flags,
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pa_stream_flags_t pa_flags,
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int latency,
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const pa_sample_spec &spec);
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SoundInputStreamInterface *ConnectInputStream(
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pa_stream *stream,
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const char *dev,
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int flags,
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pa_stream_flags_t pa_flags,
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int latency,
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const pa_sample_spec &spec);
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bool FinishOperation(pa_operation *op);
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void Lock();
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void Unlock();
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void Wait();
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void Signal();
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const char *LastError();
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pa_threaded_mainloop *mainloop_;
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pa_context *context_;
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PulseAudioSymbolTable symbol_table_;
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RTC_DISALLOW_COPY_AND_ASSIGN(PulseAudioSoundSystem);
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};
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} // namespace rtc
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#endif // HAVE_LIBPULSE
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#endif // WEBRTC_SOUND_PULSEAUDIOSOUNDSYSTEM_H_
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