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	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	#include <stropts.h>
30
31			#define MAX_QUEUE 10
32
33			int g_dsp_fd;
34	stargo	504	BOOL g_dsp_busy = False;
35	stargo	510	static BOOL g_reopened;
36			static BOOL g_swapaudio;
37			static short g_samplewidth;
38	matthewc	476
39	astrand	499	static struct audio_packet
40			{
41	matthewc	476	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	matthewc	477	char *dsp_dev = getenv("AUDIODEV");
51	matthewc	476
52	astrand	499	if (dsp_dev == NULL)
53	matthewc	477	{
54	astrand	499	dsp_dev = "/dev/audio";
55	matthewc	477	}
56
57	astrand	499	if ((g_dsp_fd = open(dsp_dev, O_WRONLY \| O_NONBLOCK)) == -1)
58	matthewc	476	{
59			perror(dsp_dev);
60			return False;
61			}
62
63			/* Non-blocking so that user interface is responsive */
64	astrand	499	fcntl(g_dsp_fd, F_SETFL, fcntl(g_dsp_fd, F_GETFL) \| O_NONBLOCK);
65
66	matthewc	476	queue_lo = queue_hi = 0;
67	stargo	510	g_reopened = True;
68	astrand	499
69	matthewc	476	return True;
70			}
71
72			void
73			wave_out_close(void)
74			{
75	matthewc	477	/* Ack all remaining packets */
76	astrand	499	while (queue_lo != queue_hi)
77	matthewc	477	{
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	stargo	538	#if defined I_FLUSH && defined FLUSHW
84	matthewc	477	/* Flush the audiobuffer */
85	astrand	499	ioctl(g_dsp_fd, I_FLUSH, FLUSHW);
86	stargo	538	#endif
87	matthewc	476	close(g_dsp_fd);
88			}
89
90			BOOL
91	astrand	499	wave_out_format_supported(WAVEFORMATEX * pwfx)
92	matthewc	476	{
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	astrand	499	wave_out_set_format(WAVEFORMATEX * pwfx)
105	matthewc	476	{
106			audio_info_t info;
107			int test = 1;
108
109			ioctl(g_dsp_fd, AUDIO_DRAIN, 0);
110	stargo	510	g_swapaudio = False;
111	matthewc	476	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	matthewc	477	/* Do we need to swap the 16bit values? (Are we BigEndian) */
122	stargo	510	g_swapaudio = !((uint8 ) (&test));
123	matthewc	476	}
124
125	stargo	510	g_samplewidth = pwfx->wBitsPerSample / 8;
126	stargo	491
127	astrand	499	if (pwfx->nChannels == 1)
128			{
129	matthewc	514	info.play.channels = 1;
130	matthewc	476	}
131	astrand	499	else if (pwfx->nChannels == 2)
132	matthewc	476	{
133	matthewc	514	info.play.channels = 2;
134	stargo	510	g_samplewidth *= 2;
135	matthewc	476	}
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	stargo	510	g_reopened = True;
143	matthewc	476
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	stargo	491	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	astrand	499	if (volume / AUDIO_MID_BALANCE != 0)
170	stargo	491	{
171	astrand	499	balance =
172			AUDIO_MID_BALANCE - (left / (volume / AUDIO_MID_BALANCE)) +
173			(right / (volume / AUDIO_MID_BALANCE));
174	stargo	491	}
175			else
176			{
177			balance = AUDIO_MID_BALANCE;
178			}
179
180	astrand	499	info.play.gain = volume / (65536 / AUDIO_MAX_GAIN);
181	stargo	491	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	matthewc	476	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	astrand	499
202	matthewc	476	queue_hi = next_hi;
203
204			packet->s = *s;
205			packet->tick = tick;
206			packet->index = index;
207	stargo	491	packet->s.p += 4;
208	matthewc	476
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	matthewc	477	static uint32 samplecnt = 0;
228			static uint32 numsamples;
229	matthewc	476
230			while (1)
231			{
232	stargo	510	if (g_reopened)
233	matthewc	477	{
234			/* Device was just (re)openend */
235			samplecnt = 0;
236			swapped = False;
237			sentcompletion = True;
238	stargo	510	g_reopened = False;
239	matthewc	477	}
240
241	matthewc	476	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	matthewc	477	/* Swap the current packet, but only once */
251	stargo	510	if (g_swapaudio && !swapped)
252	matthewc	476	{
253	astrand	499	for (i = 0; i < out->end - out->p; i += 2)
254	matthewc	476	{
255			swap = *(out->p + i);
256	stargo	491	(out->p + i) = (out->p + i + 1);
257	matthewc	476	*(out->p + i + 1) = swap;
258			}
259	stargo	491	swapped = True;
260	matthewc	476	}
261
262	astrand	499	if (sentcompletion)
263	matthewc	476	{
264			sentcompletion = False;
265	stargo	510	numsamples = (out->end - out->p) / g_samplewidth;
266	matthewc	476	}
267
268	astrand	499	len = 0;
269	matthewc	476
270	astrand	499	if (out->end != out->p)
271	matthewc	476	{
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	matthewc	477
291			/* Ack the packet, if we have played at least 70% */
292	astrand	499	if (info.play.samples >= samplecnt + ((numsamples * 7) / 10))
293	matthewc	476	{
294	matthewc	477	samplecnt += numsamples;
295	matthewc	476	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			}