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;
        }

        /* Flush the audiobuffer */
        ioctl(g_dsp_fd, I_FLUSH, FLUSHW);
        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 = AUDIO_CHANNELS_MONO;
        }
        else if (pwfx->nChannels == 2)
        {
                info.play.channels = AUDIO_CHANNELS_STEREO;
                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			/* Flush the audiobuffer */
84	astrand	499	ioctl(g_dsp_fd, I_FLUSH, FLUSHW);
85	matthewc	476	close(g_dsp_fd);
86			}
87
88			BOOL
89	astrand	499	wave_out_format_supported(WAVEFORMATEX * pwfx)
90	matthewc	476	{
91			if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
92			return False;
93			if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
94			return False;
95			if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
96			return False;
97
98			return True;
99			}
100
101			BOOL
102	astrand	499	wave_out_set_format(WAVEFORMATEX * pwfx)
103	matthewc	476	{
104			audio_info_t info;
105			int test = 1;
106
107			ioctl(g_dsp_fd, AUDIO_DRAIN, 0);
108	stargo	510	g_swapaudio = False;
109	matthewc	476	AUDIO_INITINFO(&info);
110
111
112			if (pwfx->wBitsPerSample == 8)
113			{
114			info.play.encoding = AUDIO_ENCODING_LINEAR8;
115			}
116			else if (pwfx->wBitsPerSample == 16)
117			{
118			info.play.encoding = AUDIO_ENCODING_LINEAR;
119	matthewc	477	/* Do we need to swap the 16bit values? (Are we BigEndian) */
120	stargo	510	g_swapaudio = !((uint8 ) (&test));
121	matthewc	476	}
122
123	stargo	510	g_samplewidth = pwfx->wBitsPerSample / 8;
124	stargo	491
125	astrand	499	if (pwfx->nChannels == 1)
126			{
127	matthewc	476	info.play.channels = AUDIO_CHANNELS_MONO;
128			}
129	astrand	499	else if (pwfx->nChannels == 2)
130	matthewc	476	{
131			info.play.channels = AUDIO_CHANNELS_STEREO;
132	stargo	510	g_samplewidth *= 2;
133	matthewc	476	}
134
135			info.play.sample_rate = pwfx->nSamplesPerSec;
136			info.play.precision = pwfx->wBitsPerSample;
137			info.play.samples = 0;
138			info.play.eof = 0;
139			info.play.error = 0;
140	stargo	510	g_reopened = True;
141	matthewc	476
142			if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
143			{
144			perror("AUDIO_SETINFO");
145			close(g_dsp_fd);
146			return False;
147			}
148
149			return True;
150			}
151
152			void
153	stargo	491	wave_out_volume(uint16 left, uint16 right)
154			{
155			audio_info_t info;
156			uint balance;
157			uint volume;
158
159			if (ioctl(g_dsp_fd, AUDIO_GETINFO, &info) == -1)
160			{
161			perror("AUDIO_GETINFO");
162			return;
163			}
164
165			volume = (left > right) ? left : right;
166
167	astrand	499	if (volume / AUDIO_MID_BALANCE != 0)
168	stargo	491	{
169	astrand	499	balance =
170			AUDIO_MID_BALANCE - (left / (volume / AUDIO_MID_BALANCE)) +
171			(right / (volume / AUDIO_MID_BALANCE));
172	stargo	491	}
173			else
174			{
175			balance = AUDIO_MID_BALANCE;
176			}
177
178	astrand	499	info.play.gain = volume / (65536 / AUDIO_MAX_GAIN);
179	stargo	491	info.play.balance = balance;
180
181			if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
182			{
183			perror("AUDIO_SETINFO");
184			return;
185			}
186			}
187
188			void
189	matthewc	476	wave_out_write(STREAM s, uint16 tick, uint8 index)
190			{
191			struct audio_packet *packet = &packet_queue[queue_hi];
192			unsigned int next_hi = (queue_hi + 1) % MAX_QUEUE;
193
194			if (next_hi == queue_lo)
195			{
196			error("No space to queue audio packet\n");
197			return;
198			}
199	astrand	499
200	matthewc	476	queue_hi = next_hi;
201
202			packet->s = *s;
203			packet->tick = tick;
204			packet->index = index;
205	stargo	491	packet->s.p += 4;
206	matthewc	476
207			/* we steal the data buffer from s, give it a new one */
208			s->data = malloc(s->size);
209
210			if (!g_dsp_busy)
211			wave_out_play();
212			}
213
214			void
215			wave_out_play(void)
216			{
217			struct audio_packet *packet;
218			audio_info_t info;
219			ssize_t len;
220			unsigned int i;
221			uint8 swap;
222			STREAM out;
223			static BOOL swapped = False;
224			static BOOL sentcompletion = True;
225	matthewc	477	static uint32 samplecnt = 0;
226			static uint32 numsamples;
227	matthewc	476
228			while (1)
229			{
230	stargo	510	if (g_reopened)
231	matthewc	477	{
232			/* Device was just (re)openend */
233			samplecnt = 0;
234			swapped = False;
235			sentcompletion = True;
236	stargo	510	g_reopened = False;
237	matthewc	477	}
238
239	matthewc	476	if (queue_lo == queue_hi)
240			{
241			g_dsp_busy = 0;
242			return;
243			}
244
245			packet = &packet_queue[queue_lo];
246			out = &packet->s;
247
248	matthewc	477	/* Swap the current packet, but only once */
249	stargo	510	if (g_swapaudio && !swapped)
250	matthewc	476	{
251	astrand	499	for (i = 0; i < out->end - out->p; i += 2)
252	matthewc	476	{
253			swap = *(out->p + i);
254	stargo	491	(out->p + i) = (out->p + i + 1);
255	matthewc	476	*(out->p + i + 1) = swap;
256			}
257	stargo	491	swapped = True;
258	matthewc	476	}
259
260	astrand	499	if (sentcompletion)
261	matthewc	476	{
262			sentcompletion = False;
263	stargo	510	numsamples = (out->end - out->p) / g_samplewidth;
264	matthewc	476	}
265
266	astrand	499	len = 0;
267	matthewc	476
268	astrand	499	if (out->end != out->p)
269	matthewc	476	{
270			len = write(g_dsp_fd, out->p, out->end - out->p);
271			if (len == -1)
272			{
273			if (errno != EWOULDBLOCK)
274			perror("write audio");
275			g_dsp_busy = 1;
276			return;
277			}
278			}
279
280			out->p += len;
281			if (out->p == out->end)
282			{
283			if (ioctl(g_dsp_fd, AUDIO_GETINFO, &info) == -1)
284			{
285			perror("AUDIO_GETINFO");
286			return;
287			}
288	matthewc	477
289			/* Ack the packet, if we have played at least 70% */
290	astrand	499	if (info.play.samples >= samplecnt + ((numsamples * 7) / 10))
291	matthewc	476	{
292	matthewc	477	samplecnt += numsamples;
293	matthewc	476	rdpsnd_send_completion(packet->tick, packet->index);
294			free(out->data);
295			queue_lo = (queue_lo + 1) % MAX_QUEUE;
296			swapped = False;
297			sentcompletion = True;
298			}
299			else
300			{
301			g_dsp_busy = 1;
302			return;
303			}
304			}
305			}
306			}