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