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	matthewc	476	/*
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	matthewc	477	#include <sys/audioio.h>
29	matthewc	476
30	stargo	560	#if (defined(sun) && (defined(__svr4__) \|\| defined(__SVR4)))
31			#include <stropts.h>
32			#endif
33
34	matthewc	476	#define MAX_QUEUE 10
35
36			int g_dsp_fd;
37	stargo	504	BOOL g_dsp_busy = False;
38	stargo	510	static BOOL g_reopened;
39			static BOOL g_swapaudio;
40			static short g_samplewidth;
41	matthewc	476
42	astrand	499	static struct audio_packet
43			{
44	matthewc	476	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	matthewc	477	char *dsp_dev = getenv("AUDIODEV");
54	matthewc	476
55	astrand	499	if (dsp_dev == NULL)
56	matthewc	477	{
57	astrand	499	dsp_dev = "/dev/audio";
58	matthewc	477	}
59
60	astrand	499	if ((g_dsp_fd = open(dsp_dev, O_WRONLY \| O_NONBLOCK)) == -1)
61	matthewc	476	{
62			perror(dsp_dev);
63			return False;
64			}
65
66			/* Non-blocking so that user interface is responsive */
67	astrand	499	fcntl(g_dsp_fd, F_SETFL, fcntl(g_dsp_fd, F_GETFL) \| O_NONBLOCK);
68
69	matthewc	476	queue_lo = queue_hi = 0;
70	stargo	510	g_reopened = True;
71	astrand	499
72	matthewc	476	return True;
73			}
74
75			void
76			wave_out_close(void)
77			{
78	matthewc	477	/* Ack all remaining packets */
79	astrand	499	while (queue_lo != queue_hi)
80	matthewc	477	{
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	stargo	538	#if defined I_FLUSH && defined FLUSHW
87	matthewc	477	/* Flush the audiobuffer */
88	astrand	499	ioctl(g_dsp_fd, I_FLUSH, FLUSHW);
89	stargo	538	#endif
90	matthewc	476	close(g_dsp_fd);
91			}
92
93			BOOL
94	astrand	499	wave_out_format_supported(WAVEFORMATEX * pwfx)
95	matthewc	476	{
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	astrand	499	wave_out_set_format(WAVEFORMATEX * pwfx)
108	matthewc	476	{
109			audio_info_t info;
110			int test = 1;
111
112			ioctl(g_dsp_fd, AUDIO_DRAIN, 0);
113	stargo	510	g_swapaudio = False;
114	matthewc	476	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	matthewc	477	/* Do we need to swap the 16bit values? (Are we BigEndian) */
125	stargo	510	g_swapaudio = !((uint8 ) (&test));
126	matthewc	476	}
127
128	stargo	510	g_samplewidth = pwfx->wBitsPerSample / 8;
129	stargo	491
130	astrand	499	if (pwfx->nChannels == 1)
131			{
132	matthewc	514	info.play.channels = 1;
133	matthewc	476	}
134	astrand	499	else if (pwfx->nChannels == 2)
135	matthewc	476	{
136	matthewc	514	info.play.channels = 2;
137	stargo	510	g_samplewidth *= 2;
138	matthewc	476	}
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	stargo	510	g_reopened = True;
146	matthewc	476
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	stargo	491	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	astrand	499	if (volume / AUDIO_MID_BALANCE != 0)
173	stargo	491	{
174	astrand	499	balance =
175			AUDIO_MID_BALANCE - (left / (volume / AUDIO_MID_BALANCE)) +
176			(right / (volume / AUDIO_MID_BALANCE));
177	stargo	491	}
178			else
179			{
180			balance = AUDIO_MID_BALANCE;
181			}
182
183	astrand	499	info.play.gain = volume / (65536 / AUDIO_MAX_GAIN);
184	stargo	491	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	matthewc	476	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	astrand	499
205	matthewc	476	queue_hi = next_hi;
206
207			packet->s = *s;
208			packet->tick = tick;
209			packet->index = index;
210	stargo	491	packet->s.p += 4;
211	matthewc	476
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	matthewc	477	static uint32 samplecnt = 0;
231			static uint32 numsamples;
232	matthewc	476
233			while (1)
234			{
235	stargo	510	if (g_reopened)
236	matthewc	477	{
237			/* Device was just (re)openend */
238			samplecnt = 0;
239			swapped = False;
240			sentcompletion = True;
241	stargo	510	g_reopened = False;
242	matthewc	477	}
243
244	matthewc	476	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	matthewc	477	/* Swap the current packet, but only once */
254	stargo	510	if (g_swapaudio && !swapped)
255	matthewc	476	{
256	astrand	499	for (i = 0; i < out->end - out->p; i += 2)
257	matthewc	476	{
258			swap = *(out->p + i);
259	stargo	491	(out->p + i) = (out->p + i + 1);
260	matthewc	476	*(out->p + i + 1) = swap;
261			}
262	stargo	491	swapped = True;
263	matthewc	476	}
264
265	astrand	499	if (sentcompletion)
266	matthewc	476	{
267			sentcompletion = False;
268	stargo	510	numsamples = (out->end - out->p) / g_samplewidth;
269	matthewc	476	}
270
271	astrand	499	len = 0;
272	matthewc	476
273	astrand	499	if (out->end != out->p)
274	matthewc	476	{
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	matthewc	477
294			/* Ack the packet, if we have played at least 70% */
295	astrand	499	if (info.play.samples >= samplecnt + ((numsamples * 7) / 10))
296	matthewc	476	{
297	matthewc	477	samplecnt += numsamples;
298	matthewc	476	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			}