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>

#if (defined(sun) && (defined(__svr4__) || defined(__SVR4)))
#include <stropts.h>
#endif

#define MAX_QUEUE       10

int g_dsp_fd;
BOOL g_dsp_busy = False;
static BOOL g_reopened;
static BOOL g_swapaudio;
static short g_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;
        g_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;
        }

#if defined I_FLUSH && defined FLUSHW
        /* Flush the audiobuffer */
        ioctl(g_dsp_fd, I_FLUSH, FLUSHW);
#endif
#if defined AUDIO_FLUSH
        ioctl(g_dsp_fd, AUDIO_FLUSH, NULL);
#endif
        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);
        g_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) */
                g_swapaudio = !(*(uint8 *) (&test));
        }

        g_samplewidth = pwfx->wBitsPerSample / 8;

        if (pwfx->nChannels == 1)
        {
                info.play.channels = 1;
        }
        else if (pwfx->nChannels == 2)
        {
                info.play.channels = 2;
                g_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;
        g_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 (g_reopened)
                {
                        /* Device was just (re)openend */
                        samplecnt = 0;
                        swapped = False;
                        sentcompletion = True;
                        g_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 (g_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) / g_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
30	#if (defined(sun) && (defined(__svr4__) \|\| defined(__SVR4)))
31	#include <stropts.h>
32	#endif
33
34	#define MAX_QUEUE 10
35
36	int g_dsp_fd;
37	BOOL g_dsp_busy = False;
38	static BOOL g_reopened;
39	static BOOL g_swapaudio;
40	static short g_samplewidth;
41
42	static struct audio_packet
43	{
44	struct stream s;
45	uint16 tick;
46	uint8 index;
47	} packet_queue[MAX_QUEUE];
48	static unsigned int queue_hi, queue_lo;
49
50	BOOL
51	wave_out_open(void)
52	{
53	char *dsp_dev = getenv("AUDIODEV");
54
55	if (dsp_dev == NULL)
56	{
57	dsp_dev = "/dev/audio";
58	}
59
60	if ((g_dsp_fd = open(dsp_dev, O_WRONLY \| O_NONBLOCK)) == -1)
61	{
62	perror(dsp_dev);
63	return False;
64	}
65
66	/* Non-blocking so that user interface is responsive */
67	fcntl(g_dsp_fd, F_SETFL, fcntl(g_dsp_fd, F_GETFL) \| O_NONBLOCK);
68
69	queue_lo = queue_hi = 0;
70	g_reopened = True;
71
72	return True;
73	}
74
75	void
76	wave_out_close(void)
77	{
78	/* Ack all remaining packets */
79	while (queue_lo != queue_hi)
80	{
81	rdpsnd_send_completion(packet_queue[queue_lo].tick, packet_queue[queue_lo].index);
82	free(packet_queue[queue_lo].s.data);
83	queue_lo = (queue_lo + 1) % MAX_QUEUE;
84	}
85
86	#if defined I_FLUSH && defined FLUSHW
87	/* Flush the audiobuffer */
88	ioctl(g_dsp_fd, I_FLUSH, FLUSHW);
89	#endif
90	#if defined AUDIO_FLUSH
91	ioctl(g_dsp_fd, AUDIO_FLUSH, NULL);
92	#endif
93	close(g_dsp_fd);
94	}
95
96	BOOL
97	wave_out_format_supported(WAVEFORMATEX * pwfx)
98	{
99	if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
100	return False;
101	if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
102	return False;
103	if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
104	return False;
105
106	return True;
107	}
108
109	BOOL
110	wave_out_set_format(WAVEFORMATEX * pwfx)
111	{
112	audio_info_t info;
113	int test = 1;
114
115	ioctl(g_dsp_fd, AUDIO_DRAIN, 0);
116	g_swapaudio = False;
117	AUDIO_INITINFO(&info);
118
119
120	if (pwfx->wBitsPerSample == 8)
121	{
122	info.play.encoding = AUDIO_ENCODING_LINEAR8;
123	}
124	else if (pwfx->wBitsPerSample == 16)
125	{
126	info.play.encoding = AUDIO_ENCODING_LINEAR;
127	/* Do we need to swap the 16bit values? (Are we BigEndian) */
128	g_swapaudio = !((uint8 ) (&test));
129	}
130
131	g_samplewidth = pwfx->wBitsPerSample / 8;
132
133	if (pwfx->nChannels == 1)
134	{
135	info.play.channels = 1;
136	}
137	else if (pwfx->nChannels == 2)
138	{
139	info.play.channels = 2;
140	g_samplewidth *= 2;
141	}
142
143	info.play.sample_rate = pwfx->nSamplesPerSec;
144	info.play.precision = pwfx->wBitsPerSample;
145	info.play.samples = 0;
146	info.play.eof = 0;
147	info.play.error = 0;
148	g_reopened = True;
149
150	if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
151	{
152	perror("AUDIO_SETINFO");
153	close(g_dsp_fd);
154	return False;
155	}
156
157	return True;
158	}
159
160	void
161	wave_out_volume(uint16 left, uint16 right)
162	{
163	audio_info_t info;
164	uint balance;
165	uint volume;
166
167	if (ioctl(g_dsp_fd, AUDIO_GETINFO, &info) == -1)
168	{
169	perror("AUDIO_GETINFO");
170	return;
171	}
172
173	volume = (left > right) ? left : right;
174
175	if (volume / AUDIO_MID_BALANCE != 0)
176	{
177	balance =
178	AUDIO_MID_BALANCE - (left / (volume / AUDIO_MID_BALANCE)) +
179	(right / (volume / AUDIO_MID_BALANCE));
180	}
181	else
182	{
183	balance = AUDIO_MID_BALANCE;
184	}
185
186	info.play.gain = volume / (65536 / AUDIO_MAX_GAIN);
187	info.play.balance = balance;
188
189	if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
190	{
191	perror("AUDIO_SETINFO");
192	return;
193	}
194	}
195
196	void
197	wave_out_write(STREAM s, uint16 tick, uint8 index)
198	{
199	struct audio_packet *packet = &packet_queue[queue_hi];
200	unsigned int next_hi = (queue_hi + 1) % MAX_QUEUE;
201
202	if (next_hi == queue_lo)
203	{
204	error("No space to queue audio packet\n");
205	return;
206	}
207
208	queue_hi = next_hi;
209
210	packet->s = *s;
211	packet->tick = tick;
212	packet->index = index;
213	packet->s.p += 4;
214
215	/* we steal the data buffer from s, give it a new one */
216	s->data = malloc(s->size);
217
218	if (!g_dsp_busy)
219	wave_out_play();
220	}
221
222	void
223	wave_out_play(void)
224	{
225	struct audio_packet *packet;
226	audio_info_t info;
227	ssize_t len;
228	unsigned int i;
229	uint8 swap;
230	STREAM out;
231	static BOOL swapped = False;
232	static BOOL sentcompletion = True;
233	static uint32 samplecnt = 0;
234	static uint32 numsamples;
235
236	while (1)
237	{
238	if (g_reopened)
239	{
240	/* Device was just (re)openend */
241	samplecnt = 0;
242	swapped = False;
243	sentcompletion = True;
244	g_reopened = False;
245	}
246
247	if (queue_lo == queue_hi)
248	{
249	g_dsp_busy = 0;
250	return;
251	}
252
253	packet = &packet_queue[queue_lo];
254	out = &packet->s;
255
256	/* Swap the current packet, but only once */
257	if (g_swapaudio && !swapped)
258	{
259	for (i = 0; i < out->end - out->p; i += 2)
260	{
261	swap = *(out->p + i);
262	(out->p + i) = (out->p + i + 1);
263	*(out->p + i + 1) = swap;
264	}
265	swapped = True;
266	}
267
268	if (sentcompletion)
269	{
270	sentcompletion = False;
271	numsamples = (out->end - out->p) / g_samplewidth;
272	}
273
274	len = 0;
275
276	if (out->end != out->p)
277	{
278	len = write(g_dsp_fd, out->p, out->end - out->p);
279	if (len == -1)
280	{
281	if (errno != EWOULDBLOCK)
282	perror("write audio");
283	g_dsp_busy = 1;
284	return;
285	}
286	}
287
288	out->p += len;
289	if (out->p == out->end)
290	{
291	if (ioctl(g_dsp_fd, AUDIO_GETINFO, &info) == -1)
292	{
293	perror("AUDIO_GETINFO");
294	return;
295	}
296
297	/* Ack the packet, if we have played at least 70% */
298	if (info.play.samples >= samplecnt + ((numsamples * 7) / 10))
299	{
300	samplecnt += numsamples;
301	rdpsnd_send_completion(packet->tick, packet->index);
302	free(out->data);
303	queue_lo = (queue_lo + 1) % MAX_QUEUE;
304	swapped = False;
305	sentcompletion = True;
306	}
307	else
308	{
309	g_dsp_busy = 1;
310	return;
311	}
312	}
313	}
314	}