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>

#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			#define MAX_QUEUE 10
31
32			int g_dsp_fd;
33	stargo	504	BOOL g_dsp_busy = False;
34	stargo	510	static BOOL g_reopened;
35			static BOOL g_swapaudio;
36			static short g_samplewidth;
37	matthewc	476
38	astrand	499	static struct audio_packet
39			{
40	matthewc	476	struct stream s;
41			uint16 tick;
42			uint8 index;
43			} packet_queue[MAX_QUEUE];
44			static unsigned int queue_hi, queue_lo;
45
46			BOOL
47			wave_out_open(void)
48			{
49	matthewc	477	char *dsp_dev = getenv("AUDIODEV");
50	matthewc	476
51	astrand	499	if (dsp_dev == NULL)
52	matthewc	477	{
53	astrand	499	dsp_dev = "/dev/audio";
54	matthewc	477	}
55
56	astrand	499	if ((g_dsp_fd = open(dsp_dev, O_WRONLY \| O_NONBLOCK)) == -1)
57	matthewc	476	{
58			perror(dsp_dev);
59			return False;
60			}
61
62			/* Non-blocking so that user interface is responsive */
63	astrand	499	fcntl(g_dsp_fd, F_SETFL, fcntl(g_dsp_fd, F_GETFL) \| O_NONBLOCK);
64
65	matthewc	476	queue_lo = queue_hi = 0;
66	stargo	510	g_reopened = True;
67	astrand	499
68	matthewc	476	return True;
69			}
70
71			void
72			wave_out_close(void)
73			{
74	matthewc	477	/* Ack all remaining packets */
75	astrand	499	while (queue_lo != queue_hi)
76	matthewc	477	{
77			rdpsnd_send_completion(packet_queue[queue_lo].tick, packet_queue[queue_lo].index);
78			free(packet_queue[queue_lo].s.data);
79			queue_lo = (queue_lo + 1) % MAX_QUEUE;
80			}
81
82	stargo	538	#if defined I_FLUSH && defined FLUSHW
83	matthewc	477	/* Flush the audiobuffer */
84	astrand	499	ioctl(g_dsp_fd, I_FLUSH, FLUSHW);
85	stargo	538	#endif
86	matthewc	476	close(g_dsp_fd);
87			}
88
89			BOOL
90	astrand	499	wave_out_format_supported(WAVEFORMATEX * pwfx)
91	matthewc	476	{
92			if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
93			return False;
94			if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
95			return False;
96			if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
97			return False;
98
99			return True;
100			}
101
102			BOOL
103	astrand	499	wave_out_set_format(WAVEFORMATEX * pwfx)
104	matthewc	476	{
105			audio_info_t info;
106			int test = 1;
107
108			ioctl(g_dsp_fd, AUDIO_DRAIN, 0);
109	stargo	510	g_swapaudio = False;
110	matthewc	476	AUDIO_INITINFO(&info);
111
112
113			if (pwfx->wBitsPerSample == 8)
114			{
115			info.play.encoding = AUDIO_ENCODING_LINEAR8;
116			}
117			else if (pwfx->wBitsPerSample == 16)
118			{
119			info.play.encoding = AUDIO_ENCODING_LINEAR;
120	matthewc	477	/* Do we need to swap the 16bit values? (Are we BigEndian) */
121	stargo	510	g_swapaudio = !((uint8 ) (&test));
122	matthewc	476	}
123
124	stargo	510	g_samplewidth = pwfx->wBitsPerSample / 8;
125	stargo	491
126	astrand	499	if (pwfx->nChannels == 1)
127			{
128	matthewc	514	info.play.channels = 1;
129	matthewc	476	}
130	astrand	499	else if (pwfx->nChannels == 2)
131	matthewc	476	{
132	matthewc	514	info.play.channels = 2;
133	stargo	510	g_samplewidth *= 2;
134	matthewc	476	}
135
136			info.play.sample_rate = pwfx->nSamplesPerSec;
137			info.play.precision = pwfx->wBitsPerSample;
138			info.play.samples = 0;
139			info.play.eof = 0;
140			info.play.error = 0;
141	stargo	510	g_reopened = True;
142	matthewc	476
143			if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
144			{
145			perror("AUDIO_SETINFO");
146			close(g_dsp_fd);
147			return False;
148			}
149
150			return True;
151			}
152
153			void
154	stargo	491	wave_out_volume(uint16 left, uint16 right)
155			{
156			audio_info_t info;
157			uint balance;
158			uint volume;
159
160			if (ioctl(g_dsp_fd, AUDIO_GETINFO, &info) == -1)
161			{
162			perror("AUDIO_GETINFO");
163			return;
164			}
165
166			volume = (left > right) ? left : right;
167
168	astrand	499	if (volume / AUDIO_MID_BALANCE != 0)
169	stargo	491	{
170	astrand	499	balance =
171			AUDIO_MID_BALANCE - (left / (volume / AUDIO_MID_BALANCE)) +
172			(right / (volume / AUDIO_MID_BALANCE));
173	stargo	491	}
174			else
175			{
176			balance = AUDIO_MID_BALANCE;
177			}
178
179	astrand	499	info.play.gain = volume / (65536 / AUDIO_MAX_GAIN);
180	stargo	491	info.play.balance = balance;
181
182			if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
183			{
184			perror("AUDIO_SETINFO");
185			return;
186			}
187			}
188
189			void
190	matthewc	476	wave_out_write(STREAM s, uint16 tick, uint8 index)
191			{
192			struct audio_packet *packet = &packet_queue[queue_hi];
193			unsigned int next_hi = (queue_hi + 1) % MAX_QUEUE;
194
195			if (next_hi == queue_lo)
196			{
197			error("No space to queue audio packet\n");
198			return;
199			}
200	astrand	499
201	matthewc	476	queue_hi = next_hi;
202
203			packet->s = *s;
204			packet->tick = tick;
205			packet->index = index;
206	stargo	491	packet->s.p += 4;
207	matthewc	476
208			/* we steal the data buffer from s, give it a new one */
209			s->data = malloc(s->size);
210
211			if (!g_dsp_busy)
212			wave_out_play();
213			}
214
215			void
216			wave_out_play(void)
217			{
218			struct audio_packet *packet;
219			audio_info_t info;
220			ssize_t len;
221			unsigned int i;
222			uint8 swap;
223			STREAM out;
224			static BOOL swapped = False;
225			static BOOL sentcompletion = True;
226	matthewc	477	static uint32 samplecnt = 0;
227			static uint32 numsamples;
228	matthewc	476
229			while (1)
230			{
231	stargo	510	if (g_reopened)
232	matthewc	477	{
233			/* Device was just (re)openend */
234			samplecnt = 0;
235			swapped = False;
236			sentcompletion = True;
237	stargo	510	g_reopened = False;
238	matthewc	477	}
239
240	matthewc	476	if (queue_lo == queue_hi)
241			{
242			g_dsp_busy = 0;
243			return;
244			}
245
246			packet = &packet_queue[queue_lo];
247			out = &packet->s;
248
249	matthewc	477	/* Swap the current packet, but only once */
250	stargo	510	if (g_swapaudio && !swapped)
251	matthewc	476	{
252	astrand	499	for (i = 0; i < out->end - out->p; i += 2)
253	matthewc	476	{
254			swap = *(out->p + i);
255	stargo	491	(out->p + i) = (out->p + i + 1);
256	matthewc	476	*(out->p + i + 1) = swap;
257			}
258	stargo	491	swapped = True;
259	matthewc	476	}
260
261	astrand	499	if (sentcompletion)
262	matthewc	476	{
263			sentcompletion = False;
264	stargo	510	numsamples = (out->end - out->p) / g_samplewidth;
265	matthewc	476	}
266
267	astrand	499	len = 0;
268	matthewc	476
269	astrand	499	if (out->end != out->p)
270	matthewc	476	{
271			len = write(g_dsp_fd, out->p, out->end - out->p);
272			if (len == -1)
273			{
274			if (errno != EWOULDBLOCK)
275			perror("write audio");
276			g_dsp_busy = 1;
277			return;
278			}
279			}
280
281			out->p += len;
282			if (out->p == out->end)
283			{
284			if (ioctl(g_dsp_fd, AUDIO_GETINFO, &info) == -1)
285			{
286			perror("AUDIO_GETINFO");
287			return;
288			}
289	matthewc	477
290			/* Ack the packet, if we have played at least 70% */
291	astrand	499	if (info.play.samples >= samplecnt + ((numsamples * 7) / 10))
292	matthewc	476	{
293	matthewc	477	samplecnt += numsamples;
294	matthewc	476	rdpsnd_send_completion(packet->tick, packet->index);
295			free(out->data);
296			queue_lo = (queue_lo + 1) % MAX_QUEUE;
297			swapped = False;
298			sentcompletion = True;
299			}
300			else
301			{
302			g_dsp_busy = 1;
303			return;
304			}
305			}
306			}
307			}