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;

        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) */
#ifdef B_ENDIAN
                g_swapaudio = 1;
#else
                g_swapaudio = 0;
#endif
        }

        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	stargo	561	#if defined AUDIO_FLUSH
91			ioctl(g_dsp_fd, AUDIO_FLUSH, NULL);
92			#endif
93	matthewc	476	close(g_dsp_fd);
94			}
95
96			BOOL
97	astrand	499	wave_out_format_supported(WAVEFORMATEX * pwfx)
98	matthewc	476	{
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	astrand	499	wave_out_set_format(WAVEFORMATEX * pwfx)
111	matthewc	476	{
112			audio_info_t info;
113
114			ioctl(g_dsp_fd, AUDIO_DRAIN, 0);
115	stargo	510	g_swapaudio = False;
116	matthewc	476	AUDIO_INITINFO(&info);
117
118
119			if (pwfx->wBitsPerSample == 8)
120			{
121			info.play.encoding = AUDIO_ENCODING_LINEAR8;
122			}
123			else if (pwfx->wBitsPerSample == 16)
124			{
125			info.play.encoding = AUDIO_ENCODING_LINEAR;
126	matthewc	477	/* Do we need to swap the 16bit values? (Are we BigEndian) */
127	stargo	694	#ifdef B_ENDIAN
128			g_swapaudio = 1;
129			#else
130			g_swapaudio = 0;
131			#endif
132	matthewc	476	}
133
134	stargo	510	g_samplewidth = pwfx->wBitsPerSample / 8;
135	stargo	491
136	astrand	499	if (pwfx->nChannels == 1)
137			{
138	matthewc	514	info.play.channels = 1;
139	matthewc	476	}
140	astrand	499	else if (pwfx->nChannels == 2)
141	matthewc	476	{
142	matthewc	514	info.play.channels = 2;
143	stargo	510	g_samplewidth *= 2;
144	matthewc	476	}
145
146			info.play.sample_rate = pwfx->nSamplesPerSec;
147			info.play.precision = pwfx->wBitsPerSample;
148			info.play.samples = 0;
149			info.play.eof = 0;
150			info.play.error = 0;
151	stargo	510	g_reopened = True;
152	matthewc	476
153			if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
154			{
155			perror("AUDIO_SETINFO");
156			close(g_dsp_fd);
157			return False;
158			}
159
160			return True;
161			}
162
163			void
164	stargo	491	wave_out_volume(uint16 left, uint16 right)
165			{
166			audio_info_t info;
167			uint balance;
168			uint volume;
169
170			if (ioctl(g_dsp_fd, AUDIO_GETINFO, &info) == -1)
171			{
172			perror("AUDIO_GETINFO");
173			return;
174			}
175
176			volume = (left > right) ? left : right;
177
178	astrand	499	if (volume / AUDIO_MID_BALANCE != 0)
179	stargo	491	{
180	astrand	499	balance =
181			AUDIO_MID_BALANCE - (left / (volume / AUDIO_MID_BALANCE)) +
182			(right / (volume / AUDIO_MID_BALANCE));
183	stargo	491	}
184			else
185			{
186			balance = AUDIO_MID_BALANCE;
187			}
188
189	astrand	499	info.play.gain = volume / (65536 / AUDIO_MAX_GAIN);
190	stargo	491	info.play.balance = balance;
191
192			if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
193			{
194			perror("AUDIO_SETINFO");
195			return;
196			}
197			}
198
199			void
200	matthewc	476	wave_out_write(STREAM s, uint16 tick, uint8 index)
201			{
202			struct audio_packet *packet = &packet_queue[queue_hi];
203			unsigned int next_hi = (queue_hi + 1) % MAX_QUEUE;
204
205			if (next_hi == queue_lo)
206			{
207			error("No space to queue audio packet\n");
208			return;
209			}
210	astrand	499
211	matthewc	476	queue_hi = next_hi;
212
213			packet->s = *s;
214			packet->tick = tick;
215			packet->index = index;
216	stargo	491	packet->s.p += 4;
217	matthewc	476
218			/* we steal the data buffer from s, give it a new one */
219			s->data = malloc(s->size);
220
221			if (!g_dsp_busy)
222			wave_out_play();
223			}
224
225			void
226			wave_out_play(void)
227			{
228			struct audio_packet *packet;
229			audio_info_t info;
230			ssize_t len;
231			unsigned int i;
232			uint8 swap;
233			STREAM out;
234			static BOOL swapped = False;
235			static BOOL sentcompletion = True;
236	matthewc	477	static uint32 samplecnt = 0;
237			static uint32 numsamples;
238	matthewc	476
239			while (1)
240			{
241	stargo	510	if (g_reopened)
242	matthewc	477	{
243			/* Device was just (re)openend */
244			samplecnt = 0;
245			swapped = False;
246			sentcompletion = True;
247	stargo	510	g_reopened = False;
248	matthewc	477	}
249
250	matthewc	476	if (queue_lo == queue_hi)
251			{
252			g_dsp_busy = 0;
253			return;
254			}
255
256			packet = &packet_queue[queue_lo];
257			out = &packet->s;
258
259	matthewc	477	/* Swap the current packet, but only once */
260	stargo	510	if (g_swapaudio && !swapped)
261	matthewc	476	{
262	astrand	499	for (i = 0; i < out->end - out->p; i += 2)
263	matthewc	476	{
264			swap = *(out->p + i);
265	stargo	491	(out->p + i) = (out->p + i + 1);
266	matthewc	476	*(out->p + i + 1) = swap;
267			}
268	stargo	491	swapped = True;
269	matthewc	476	}
270
271	astrand	499	if (sentcompletion)
272	matthewc	476	{
273			sentcompletion = False;
274	stargo	510	numsamples = (out->end - out->p) / g_samplewidth;
275	matthewc	476	}
276
277	astrand	499	len = 0;
278	matthewc	476
279	astrand	499	if (out->end != out->p)
280	matthewc	476	{
281			len = write(g_dsp_fd, out->p, out->end - out->p);
282			if (len == -1)
283			{
284			if (errno != EWOULDBLOCK)
285			perror("write audio");
286			g_dsp_busy = 1;
287			return;
288			}
289			}
290
291			out->p += len;
292			if (out->p == out->end)
293			{
294			if (ioctl(g_dsp_fd, AUDIO_GETINFO, &info) == -1)
295			{
296			perror("AUDIO_GETINFO");
297			return;
298			}
299	matthewc	477
300			/* Ack the packet, if we have played at least 70% */
301	astrand	499	if (info.play.samples >= samplecnt + ((numsamples * 7) / 10))
302	matthewc	476	{
303	matthewc	477	samplecnt += numsamples;
304	matthewc	476	rdpsnd_send_completion(packet->tick, packet->index);
305			free(out->data);
306			queue_lo = (queue_lo + 1) % MAX_QUEUE;
307			swapped = False;
308			sentcompletion = True;
309			}
310			else
311			{
312			g_dsp_busy = 1;
313			return;
314			}
315			}
316			}
317			}