trunk/rdesktop/rdpsnd_sun.c

/* 
   rdesktop: A Remote Desktop Protocol client.
   Sound Channel Process Functions - Sun
   Copyright (C) Matthew Chapman 2003
   Copyright (C) GuoJunBo guojunbo@ict.ac.cn 2003
   Copyright (C) Michael Gernoth mike@zerfleddert.de 2003

   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., 675 Mass Ave, Cambridge, MA 02139, USA.
*/

#include "rdesktop.h"
#include <unistd.h>
#include <fcntl.h>
#include <errno.h>
#include <sys/ioctl.h>
#include <sys/audioio.h>
#include <stropts.h>

#define MAX_QUEUE       10

int g_dsp_fd;
BOOL g_dsp_busy;
static BOOL reopened;
static BOOL swapaudio;
static short samplewidth;

static struct audio_packet {
        struct stream s;
        uint16 tick;
        uint8 index;
} packet_queue[MAX_QUEUE];
static unsigned int queue_hi, queue_lo;

BOOL
wave_out_open(void)
{
        char *dsp_dev = getenv("AUDIODEV");

        if ( dsp_dev == NULL )
        {
                dsp_dev="/dev/audio";
        }

        if ((g_dsp_fd = open(dsp_dev, O_WRONLY|O_NONBLOCK)) == -1)
        {
                perror(dsp_dev);
                return False;
        }

        /* Non-blocking so that user interface is responsive */
        fcntl(g_dsp_fd, F_SETFL, fcntl(g_dsp_fd, F_GETFL)|O_NONBLOCK);
        
        queue_lo = queue_hi = 0;
        reopened = True;
        
        return True;
}

void
wave_out_close(void)
{
        /* Ack all remaining packets */
        while ( queue_lo != queue_hi )
        {
                rdpsnd_send_completion(packet_queue[queue_lo].tick, packet_queue[queue_lo].index);
                free(packet_queue[queue_lo].s.data);
                queue_lo = (queue_lo + 1) % MAX_QUEUE;
        }

        /* Flush the audiobuffer */
        ioctl(g_dsp_fd,I_FLUSH,FLUSHW);
        close(g_dsp_fd);
}

BOOL
wave_out_format_supported(WAVEFORMATEX *pwfx)
{
        if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
                return False;
        if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
                return False;
        if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
                return False;

        return True;
}

BOOL
wave_out_set_format(WAVEFORMATEX *pwfx)
{
        audio_info_t info;
        int test = 1;

        ioctl(g_dsp_fd, AUDIO_DRAIN, 0);
        swapaudio = False;
        AUDIO_INITINFO(&info);


        if (pwfx->wBitsPerSample == 8)
        {
                info.play.encoding = AUDIO_ENCODING_LINEAR8;
        }
        else if (pwfx->wBitsPerSample == 16)
        {
                info.play.encoding = AUDIO_ENCODING_LINEAR;
                /* Do we need to swap the 16bit values? (Are we BigEndian) */
                swapaudio = !(*(uint8 *) (&test));
        }

        samplewidth = pwfx->wBitsPerSample/8;

        if (pwfx->nChannels == 1 )
        {       
                info.play.channels = AUDIO_CHANNELS_MONO;
        }
        else if (pwfx->nChannels == 2 )
        {
                info.play.channels = AUDIO_CHANNELS_STEREO;
                samplewidth *= 2;
        }

        info.play.sample_rate = pwfx->nSamplesPerSec;
        info.play.precision = pwfx->wBitsPerSample;
        info.play.samples = 0;
        info.play.eof = 0;
        info.play.error = 0;
        reopened = True;

        if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
        {
                perror("AUDIO_SETINFO");
                close(g_dsp_fd);
                return False;
        }

        return True;
}

void
wave_out_volume(uint16 left, uint16 right)
{
        audio_info_t info;
        uint balance;
        uint volume;

        if (ioctl(g_dsp_fd, AUDIO_GETINFO, &info) == -1)
        {
                perror("AUDIO_GETINFO");
                return;
        }

        volume = (left > right) ? left : right;

        if ( volume/AUDIO_MID_BALANCE != 0 )
        {
                balance = AUDIO_MID_BALANCE - (left/(volume/AUDIO_MID_BALANCE)) + (right/(volume/AUDIO_MID_BALANCE));
        }
        else
        {
                balance = AUDIO_MID_BALANCE;
        }

        info.play.gain = volume/(65536/AUDIO_MAX_GAIN);
        info.play.balance = balance;

        if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
        {
                perror("AUDIO_SETINFO");
                return;
        }
}

void
wave_out_write(STREAM s, uint16 tick, uint8 index)
{
        struct audio_packet *packet = &packet_queue[queue_hi];
        unsigned int next_hi = (queue_hi + 1) % MAX_QUEUE;

        if (next_hi == queue_lo)
        {
                error("No space to queue audio packet\n");
                return;
        }
        
        queue_hi = next_hi;

        packet->s = *s;
        packet->tick = tick;
        packet->index = index;
        packet->s.p += 4;

        /* we steal the data buffer from s, give it a new one */
        s->data = malloc(s->size);

        if (!g_dsp_busy)
                wave_out_play();
}

void
wave_out_play(void)
{
        struct audio_packet *packet;
        audio_info_t info;
        ssize_t len;
        unsigned int i;
        uint8 swap;
        STREAM out;
        static BOOL swapped = False;
        static BOOL sentcompletion = True;
        static uint32 samplecnt = 0;
        static uint32 numsamples;

        while (1)
        {
                if ( reopened )
                {
                        /* Device was just (re)openend */
                        samplecnt = 0;
                        swapped = False;
                        sentcompletion = True;
                        reopened = False;
                }

                if (queue_lo == queue_hi)
                {
                        g_dsp_busy = 0;
                        return;
                }

                packet = &packet_queue[queue_lo];
                out = &packet->s;

                /* Swap the current packet, but only once */
                if ( swapaudio && ! swapped )
                {
                        for ( i = 0; i < out->end - out->p; i += 2 )
                        {
                                swap = *(out->p + i);
                                *(out->p + i) = *(out->p + i + 1);
                                *(out->p + i + 1) = swap;
                        }
                        swapped = True;
                }

                if ( sentcompletion )
                {
                        sentcompletion = False;
                        numsamples = (out->end - out->p)/samplewidth;
                }

                len=0;

                if ( out->end != out->p )
                {
                        len = write(g_dsp_fd, out->p, out->end - out->p);
                        if (len == -1)
                        {
                                if (errno != EWOULDBLOCK)
                                        perror("write audio");
                                g_dsp_busy = 1;
                                return;
                        }
                }

                out->p += len;
                if (out->p == out->end)
                {
                        if (ioctl(g_dsp_fd, AUDIO_GETINFO, &info) == -1)
                        {
                                perror("AUDIO_GETINFO");
                                return;
                        }

                        /* Ack the packet, if we have played at least 70% */
                        if ( info.play.samples >= samplecnt+((numsamples*7)/10) )
                        {
                                samplecnt += numsamples;
                                rdpsnd_send_completion(packet->tick, packet->index);
                                free(out->data);
                                queue_lo = (queue_lo + 1) % MAX_QUEUE;
                                swapped = False;
                                sentcompletion = True;
                        }
                        else
                        {
                                g_dsp_busy = 1;
                                return;
                        }
                }
        }
}
1	/*
2	rdesktop: A Remote Desktop Protocol client.
3	Sound Channel Process Functions - Sun
4	Copyright (C) Matthew Chapman 2003
5	Copyright (C) GuoJunBo guojunbo@ict.ac.cn 2003
6	Copyright (C) Michael Gernoth mike@zerfleddert.de 2003
7
8	This program is free software; you can redistribute it and/or modify
9	it under the terms of the GNU General Public License as published by
10	the Free Software Foundation; either version 2 of the License, or
11	(at your option) any later version.
12
13	This program is distributed in the hope that it will be useful,
14	but WITHOUT ANY WARRANTY; without even the implied warranty of
15	MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
16	GNU General Public License for more details.
17
18	You should have received a copy of the GNU General Public License
19	along with this program; if not, write to the Free Software
20	Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
21	*/
22
23	#include "rdesktop.h"
24	#include <unistd.h>
25	#include <fcntl.h>
26	#include <errno.h>
27	#include <sys/ioctl.h>
28	#include <sys/audioio.h>
29	#include <stropts.h>
30
31	#define MAX_QUEUE 10
32
33	int g_dsp_fd;
34	BOOL g_dsp_busy;
35	static BOOL reopened;
36	static BOOL swapaudio;
37	static short samplewidth;
38
39	static struct audio_packet {
40	struct stream s;
41	uint16 tick;
42	uint8 index;
43	} packet_queue[MAX_QUEUE];
44	static unsigned int queue_hi, queue_lo;
45
46	BOOL
47	wave_out_open(void)
48	{
49	char *dsp_dev = getenv("AUDIODEV");
50
51	if ( dsp_dev == NULL )
52	{
53	dsp_dev="/dev/audio";
54	}
55
56	if ((g_dsp_fd = open(dsp_dev, O_WRONLY\|O_NONBLOCK)) == -1)
57	{
58	perror(dsp_dev);
59	return False;
60	}
61
62	/* Non-blocking so that user interface is responsive */
63	fcntl(g_dsp_fd, F_SETFL, fcntl(g_dsp_fd, F_GETFL)\|O_NONBLOCK);
64
65	queue_lo = queue_hi = 0;
66	reopened = True;
67
68	return True;
69	}
70
71	void
72	wave_out_close(void)
73	{
74	/* Ack all remaining packets */
75	while ( queue_lo != queue_hi )
76	{
77	rdpsnd_send_completion(packet_queue[queue_lo].tick, packet_queue[queue_lo].index);
78	free(packet_queue[queue_lo].s.data);
79	queue_lo = (queue_lo + 1) % MAX_QUEUE;
80	}
81
82	/* Flush the audiobuffer */
83	ioctl(g_dsp_fd,I_FLUSH,FLUSHW);
84	close(g_dsp_fd);
85	}
86
87	BOOL
88	wave_out_format_supported(WAVEFORMATEX *pwfx)
89	{
90	if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
91	return False;
92	if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
93	return False;
94	if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
95	return False;
96
97	return True;
98	}
99
100	BOOL
101	wave_out_set_format(WAVEFORMATEX *pwfx)
102	{
103	audio_info_t info;
104	int test = 1;
105
106	ioctl(g_dsp_fd, AUDIO_DRAIN, 0);
107	swapaudio = False;
108	AUDIO_INITINFO(&info);
109
110
111	if (pwfx->wBitsPerSample == 8)
112	{
113	info.play.encoding = AUDIO_ENCODING_LINEAR8;
114	}
115	else if (pwfx->wBitsPerSample == 16)
116	{
117	info.play.encoding = AUDIO_ENCODING_LINEAR;
118	/* Do we need to swap the 16bit values? (Are we BigEndian) */
119	swapaudio = !((uint8 ) (&test));
120	}
121
122	samplewidth = pwfx->wBitsPerSample/8;
123
124	if (pwfx->nChannels == 1 )
125	{
126	info.play.channels = AUDIO_CHANNELS_MONO;
127	}
128	else if (pwfx->nChannels == 2 )
129	{
130	info.play.channels = AUDIO_CHANNELS_STEREO;
131	samplewidth *= 2;
132	}
133
134	info.play.sample_rate = pwfx->nSamplesPerSec;
135	info.play.precision = pwfx->wBitsPerSample;
136	info.play.samples = 0;
137	info.play.eof = 0;
138	info.play.error = 0;
139	reopened = True;
140
141	if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
142	{
143	perror("AUDIO_SETINFO");
144	close(g_dsp_fd);
145	return False;
146	}
147
148	return True;
149	}
150
151	void
152	wave_out_volume(uint16 left, uint16 right)
153	{
154	audio_info_t info;
155	uint balance;
156	uint volume;
157
158	if (ioctl(g_dsp_fd, AUDIO_GETINFO, &info) == -1)
159	{
160	perror("AUDIO_GETINFO");
161	return;
162	}
163
164	volume = (left > right) ? left : right;
165
166	if ( volume/AUDIO_MID_BALANCE != 0 )
167	{
168	balance = AUDIO_MID_BALANCE - (left/(volume/AUDIO_MID_BALANCE)) + (right/(volume/AUDIO_MID_BALANCE));
169	}
170	else
171	{
172	balance = AUDIO_MID_BALANCE;
173	}
174
175	info.play.gain = volume/(65536/AUDIO_MAX_GAIN);
176	info.play.balance = balance;
177
178	if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
179	{
180	perror("AUDIO_SETINFO");
181	return;
182	}
183	}
184
185	void
186	wave_out_write(STREAM s, uint16 tick, uint8 index)
187	{
188	struct audio_packet *packet = &packet_queue[queue_hi];
189	unsigned int next_hi = (queue_hi + 1) % MAX_QUEUE;
190
191	if (next_hi == queue_lo)
192	{
193	error("No space to queue audio packet\n");
194	return;
195	}
196
197	queue_hi = next_hi;
198
199	packet->s = *s;
200	packet->tick = tick;
201	packet->index = index;
202	packet->s.p += 4;
203
204	/* we steal the data buffer from s, give it a new one */
205	s->data = malloc(s->size);
206
207	if (!g_dsp_busy)
208	wave_out_play();
209	}
210
211	void
212	wave_out_play(void)
213	{
214	struct audio_packet *packet;
215	audio_info_t info;
216	ssize_t len;
217	unsigned int i;
218	uint8 swap;
219	STREAM out;
220	static BOOL swapped = False;
221	static BOOL sentcompletion = True;
222	static uint32 samplecnt = 0;
223	static uint32 numsamples;
224
225	while (1)
226	{
227	if ( reopened )
228	{
229	/* Device was just (re)openend */
230	samplecnt = 0;
231	swapped = False;
232	sentcompletion = True;
233	reopened = False;
234	}
235
236	if (queue_lo == queue_hi)
237	{
238	g_dsp_busy = 0;
239	return;
240	}
241
242	packet = &packet_queue[queue_lo];
243	out = &packet->s;
244
245	/* Swap the current packet, but only once */
246	if ( swapaudio && ! swapped )
247	{
248	for ( i = 0; i < out->end - out->p; i += 2 )
249	{
250	swap = *(out->p + i);
251	(out->p + i) = (out->p + i + 1);
252	*(out->p + i + 1) = swap;
253	}
254	swapped = True;
255	}
256
257	if ( sentcompletion )
258	{
259	sentcompletion = False;
260	numsamples = (out->end - out->p)/samplewidth;
261	}
262
263	len=0;
264
265	if ( out->end != out->p )
266	{
267	len = write(g_dsp_fd, out->p, out->end - out->p);
268	if (len == -1)
269	{
270	if (errno != EWOULDBLOCK)
271	perror("write audio");
272	g_dsp_busy = 1;
273	return;
274	}
275	}
276
277	out->p += len;
278	if (out->p == out->end)
279	{
280	if (ioctl(g_dsp_fd, AUDIO_GETINFO, &info) == -1)
281	{
282	perror("AUDIO_GETINFO");
283	return;
284	}
285
286	/* Ack the packet, if we have played at least 70% */
287	if ( info.play.samples >= samplecnt+((numsamples*7)/10) )
288	{
289	samplecnt += numsamples;
290	rdpsnd_send_completion(packet->tick, packet->index);
291	free(out->data);
292	queue_lo = (queue_lo + 1) % MAX_QUEUE;
293	swapped = False;
294	sentcompletion = True;
295	}
296	else
297	{
298	g_dsp_busy = 1;
299	return;
300	}
301	}
302	}
303	}