From e2bc6f4153813cc570ae814c8ddb74628009b488 Mon Sep 17 00:00:00 2001 From: Michal Kubecek Date: Mon, 13 Apr 2015 09:21:39 +0200 Subject: initial checkin Check in contents of upstream 1.4.2 tarball, exclude generated files. --- src/audio/audio_device.h | 126 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 126 insertions(+) create mode 100644 src/audio/audio_device.h (limited to 'src/audio/audio_device.h') diff --git a/src/audio/audio_device.h b/src/audio/audio_device.h new file mode 100644 index 0000000..27503f8 --- /dev/null +++ b/src/audio/audio_device.h @@ -0,0 +1,126 @@ +/* + Copyright (C) 2005-2009 Michel de Boer + + This program is free software; you can redistribute it and/or modify + it under the terms of the GNU General Public License as published by + the Free Software Foundation; either version 2 of the License, or + (at your option) any later version. + + This program is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + GNU General Public License for more details. + + You should have received a copy of the GNU General Public License + along with this program; if not, write to the Free Software + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA +*/ + +#ifndef _AUDIO_DEVICE_H +#define _AUDIO_DEVICE_H + +#include +#include "twinkle_config.h" + +using namespace std; + +#ifndef _SYS_SETTINGS_H +class t_audio_device; +#endif + +enum t_audio_sampleformat { + SAMPLEFORMAT_U8, + SAMPLEFORMAT_S8, + SAMPLEFORMAT_S16, + SAMPLEFORMAT_U16 +}; + +class t_audio_io { +public: + virtual ~t_audio_io(); + virtual void enable(bool enable_playback, bool enable_recording) = 0; + virtual void flush(bool playback_buffer, bool recording_buffer) = 0; + // Returns the number of bytes that can be written/read without blocking + virtual int get_buffer_space(bool is_recording_buffer) = 0; + // Returns the size of the hardware buffer + virtual int get_buffer_size(bool is_recording_buffer) = 0; + + /** Check if a play buffer underrun occurred. */ + virtual bool play_buffer_underrun(void) = 0; + + virtual int read(unsigned char* buf, int len) = 0; + virtual int write(const unsigned char* buf, int len) = 0; + virtual int get_sample_rate(void) const; + + static t_audio_io* open(const t_audio_device& dev, bool playback, + bool capture, bool blocking, int channels, t_audio_sampleformat format, + int sample_rate, bool short_latency); + + // Validate if an audio device can be opened. + static bool validate(const t_audio_device& dev, bool playback, bool capture); + +protected: + virtual bool open(const string& device, bool playback, bool capture, + bool blocking, int channels, t_audio_sampleformat format, + int sample_rate, bool short_latency); + +private: + int _sample_rate; + +}; + +class t_oss_io : public t_audio_io { +public: + t_oss_io(); + virtual ~t_oss_io(); + void enable(bool enable_playback, bool enable_recording); + void flush(bool playback_buffer, bool recording_buffer); + int get_buffer_space(bool is_recording_buffer); + int get_buffer_size(bool is_recording_buffer); + bool play_buffer_underrun(void); + int read(unsigned char* buf, int len); + int write(const unsigned char* buf, int len); +protected: + bool open(const string& device, bool playback, bool capture, bool blocking, + int channels, t_audio_sampleformat format, int sample_rate, + bool short_latency); +private: + int fd; + int play_buffersize, rec_buffersize; +}; + +#ifdef HAVE_LIBASOUND +class t_alsa_io : public t_audio_io { +public: + t_alsa_io(); + virtual ~t_alsa_io(); + void enable(bool enable_playback, bool enable_recording); + void flush(bool playback_buffer, bool recording_buffer); + int get_buffer_space(bool is_recording_buffer); + int get_buffer_size(bool is_recording_buffer); + bool play_buffer_underrun(void); + int read(unsigned char* buf, int len); + int write(const unsigned char* buf, int len); +protected: + bool open(const string& device, bool playback, bool capture, bool blocking, + int channels, t_audio_sampleformat format, int sample_rate, + bool short_latency); +private: + struct _snd_pcm *pcm_play_ptr, *pcm_rec_ptr; + int play_framesize, rec_framesize; + int play_buffersize, rec_buffersize; + int play_periods, rec_periods; + bool short_latency; + + // snd_pcm_delay should return the number of bytes in the buffer. + // For some reason however, if the capture device is a software mixer, + // it returns inaccurate values. + // This flag if the functionality is broken. + bool rec_delay_broken; + + // Indicates if snd_pcm_pause works for this device + bool can_pause; +}; +#endif + +#endif -- cgit v1.2.3