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
        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	close(g_dsp_fd);
91	}
92
93	BOOL
94	wave_out_format_supported(WAVEFORMATEX * pwfx)
95	{
96	if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
97	return False;
98	if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
99	return False;
100	if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
101	return False;
102
103	return True;
104	}
105
106	BOOL
107	wave_out_set_format(WAVEFORMATEX * pwfx)
108	{
109	audio_info_t info;
110	int test = 1;
111
112	ioctl(g_dsp_fd, AUDIO_DRAIN, 0);
113	g_swapaudio = False;
114	AUDIO_INITINFO(&info);
115
116
117	if (pwfx->wBitsPerSample == 8)
118	{
119	info.play.encoding = AUDIO_ENCODING_LINEAR8;
120	}
121	else if (pwfx->wBitsPerSample == 16)
122	{
123	info.play.encoding = AUDIO_ENCODING_LINEAR;
124	/* Do we need to swap the 16bit values? (Are we BigEndian) */
125	g_swapaudio = !((uint8 ) (&test));
126	}
127
128	g_samplewidth = pwfx->wBitsPerSample / 8;
129
130	if (pwfx->nChannels == 1)
131	{
132	info.play.channels = 1;
133	}
134	else if (pwfx->nChannels == 2)
135	{
136	info.play.channels = 2;
137	g_samplewidth *= 2;
138	}
139
140	info.play.sample_rate = pwfx->nSamplesPerSec;
141	info.play.precision = pwfx->wBitsPerSample;
142	info.play.samples = 0;
143	info.play.eof = 0;
144	info.play.error = 0;
145	g_reopened = True;
146
147	if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
148	{
149	perror("AUDIO_SETINFO");
150	close(g_dsp_fd);
151	return False;
152	}
153
154	return True;
155	}
156
157	void
158	wave_out_volume(uint16 left, uint16 right)
159	{
160	audio_info_t info;
161	uint balance;
162	uint volume;
163
164	if (ioctl(g_dsp_fd, AUDIO_GETINFO, &info) == -1)
165	{
166	perror("AUDIO_GETINFO");
167	return;
168	}
169
170	volume = (left > right) ? left : right;
171
172	if (volume / AUDIO_MID_BALANCE != 0)
173	{
174	balance =
175	AUDIO_MID_BALANCE - (left / (volume / AUDIO_MID_BALANCE)) +
176	(right / (volume / AUDIO_MID_BALANCE));
177	}
178	else
179	{
180	balance = AUDIO_MID_BALANCE;
181	}
182
183	info.play.gain = volume / (65536 / AUDIO_MAX_GAIN);
184	info.play.balance = balance;
185
186	if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
187	{
188	perror("AUDIO_SETINFO");
189	return;
190	}
191	}
192
193	void
194	wave_out_write(STREAM s, uint16 tick, uint8 index)
195	{
196	struct audio_packet *packet = &packet_queue[queue_hi];
197	unsigned int next_hi = (queue_hi + 1) % MAX_QUEUE;
198
199	if (next_hi == queue_lo)
200	{
201	error("No space to queue audio packet\n");
202	return;
203	}
204
205	queue_hi = next_hi;
206
207	packet->s = *s;
208	packet->tick = tick;
209	packet->index = index;
210	packet->s.p += 4;
211
212	/* we steal the data buffer from s, give it a new one */
213	s->data = malloc(s->size);
214
215	if (!g_dsp_busy)
216	wave_out_play();
217	}
218
219	void
220	wave_out_play(void)
221	{
222	struct audio_packet *packet;
223	audio_info_t info;
224	ssize_t len;
225	unsigned int i;
226	uint8 swap;
227	STREAM out;
228	static BOOL swapped = False;
229	static BOOL sentcompletion = True;
230	static uint32 samplecnt = 0;
231	static uint32 numsamples;
232
233	while (1)
234	{
235	if (g_reopened)
236	{
237	/* Device was just (re)openend */
238	samplecnt = 0;
239	swapped = False;
240	sentcompletion = True;
241	g_reopened = False;
242	}
243
244	if (queue_lo == queue_hi)
245	{
246	g_dsp_busy = 0;
247	return;
248	}
249
250	packet = &packet_queue[queue_lo];
251	out = &packet->s;
252
253	/* Swap the current packet, but only once */
254	if (g_swapaudio && !swapped)
255	{
256	for (i = 0; i < out->end - out->p; i += 2)
257	{
258	swap = *(out->p + i);
259	(out->p + i) = (out->p + i + 1);
260	*(out->p + i + 1) = swap;
261	}
262	swapped = True;
263	}
264
265	if (sentcompletion)
266	{
267	sentcompletion = False;
268	numsamples = (out->end - out->p) / g_samplewidth;
269	}
270
271	len = 0;
272
273	if (out->end != out->p)
274	{
275	len = write(g_dsp_fd, out->p, out->end - out->p);
276	if (len == -1)
277	{
278	if (errno != EWOULDBLOCK)
279	perror("write audio");
280	g_dsp_busy = 1;
281	return;
282	}
283	}
284
285	out->p += len;
286	if (out->p == out->end)
287	{
288	if (ioctl(g_dsp_fd, AUDIO_GETINFO, &info) == -1)
289	{
290	perror("AUDIO_GETINFO");
291	return;
292	}
293
294	/* Ack the packet, if we have played at least 70% */
295	if (info.play.samples >= samplecnt + ((numsamples * 7) / 10))
296	{
297	samplecnt += numsamples;
298	rdpsnd_send_completion(packet->tick, packet->index);
299	free(out->data);
300	queue_lo = (queue_lo + 1) % MAX_QUEUE;
301	swapped = False;
302	sentcompletion = True;
303	}
304	else
305	{
306	g_dsp_busy = 1;
307	return;
308	}
309	}
310	}
311	}