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authorMichal Kubecek <mkubecek@suse.cz>2015-04-13 09:21:39 +0200
committerMichal Kubecek <mkubecek@suse.cz>2015-04-13 09:21:39 +0200
commite2bc6f4153813cc570ae814c8ddb74628009b488 (patch)
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initial checkin
Check in contents of upstream 1.4.2 tarball, exclude generated files.
Diffstat (limited to 'src/audio/audio_device.h')
-rw-r--r--src/audio/audio_device.h126
1 files changed, 126 insertions, 0 deletions
diff --git a/src/audio/audio_device.h b/src/audio/audio_device.h
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+/*
+ Copyright (C) 2005-2009 Michel de Boer <michel@twinklephone.com>
+
+ 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 <string>
+#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