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;
static BOOL reopened;
static BOOL swapaudio;
static short 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;
        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);
        swapaudio = False;
        AUDIO_INITINFO(&info);


        if (pwfx->wBitsPerSample == 8)
        {
                info.play.encoding = AUDIO_ENCODING_LINEAR8;
                samplewidth = 1;
        }
        else if (pwfx->wBitsPerSample == 16)
        {
                info.play.encoding = AUDIO_ENCODING_LINEAR;
                samplewidth = 2;
                /* Do we need to swap the 16bit values? (Are we BigEndian) */
                swapaudio = !(*(uint8 *) (&test));
        }

        if (pwfx->nChannels == 1 )
        {       
                info.play.channels = AUDIO_CHANNELS_MONO;
        }
        else if (pwfx->nChannels == 2 )
        {
                info.play.channels = AUDIO_CHANNELS_STEREO;
                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;

        if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
        {
                perror("AUDIO_SETINFO");
                close(g_dsp_fd);
                return False;
        }

        return True;
}

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;

        /* 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 ( reopened )
                {
                        /* Device was just (re)openend */
                        samplecnt = 0;
                        swapped = False;
                        sentcompletion = True;
                        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 ( 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)/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	/*
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	#include <sys/audioio.h>
29	#include <stropts.h>
30
31	#define MAX_QUEUE 10
32
33	int g_dsp_fd;
34	BOOL g_dsp_busy;
35	static BOOL reopened;
36	static BOOL swapaudio;
37	static short samplewidth;
38
39	static struct audio_packet {
40	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	char *dsp_dev = getenv("AUDIODEV");
50
51	if ( dsp_dev == NULL )
52	{
53	dsp_dev="/dev/audio";
54	}
55
56	if ((g_dsp_fd = open(dsp_dev, O_WRONLY\|O_NONBLOCK)) == -1)
57	{
58	perror(dsp_dev);
59	return False;
60	}
61
62	/* Non-blocking so that user interface is responsive */
63	fcntl(g_dsp_fd, F_SETFL, fcntl(g_dsp_fd, F_GETFL)\|O_NONBLOCK);
64
65	queue_lo = queue_hi = 0;
66	reopened = True;
67
68	return True;
69	}
70
71	void
72	wave_out_close(void)
73	{
74	/* Ack all remaining packets */
75	while ( queue_lo != queue_hi )
76	{
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	/* Flush the audiobuffer */
83	ioctl(g_dsp_fd,I_FLUSH,FLUSHW);
84	close(g_dsp_fd);
85	}
86
87	BOOL
88	wave_out_format_supported(WAVEFORMATEX *pwfx)
89	{
90	if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
91	return False;
92	if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
93	return False;
94	if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
95	return False;
96
97	return True;
98	}
99
100	BOOL
101	wave_out_set_format(WAVEFORMATEX *pwfx)
102	{
103	audio_info_t info;
104	int test = 1;
105
106	ioctl(g_dsp_fd, AUDIO_DRAIN, 0);
107	swapaudio = False;
108	AUDIO_INITINFO(&info);
109
110
111	if (pwfx->wBitsPerSample == 8)
112	{
113	info.play.encoding = AUDIO_ENCODING_LINEAR8;
114	samplewidth = 1;
115	}
116	else if (pwfx->wBitsPerSample == 16)
117	{
118	info.play.encoding = AUDIO_ENCODING_LINEAR;
119	samplewidth = 2;
120	/* Do we need to swap the 16bit values? (Are we BigEndian) */
121	swapaudio = !((uint8 ) (&test));
122	}
123
124	if (pwfx->nChannels == 1 )
125	{
126	info.play.channels = AUDIO_CHANNELS_MONO;
127	}
128	else if (pwfx->nChannels == 2 )
129	{
130	info.play.channels = AUDIO_CHANNELS_STEREO;
131	samplewidth *= 2;
132	}
133
134	info.play.sample_rate = pwfx->nSamplesPerSec;
135	info.play.precision = pwfx->wBitsPerSample;
136	info.play.samples = 0;
137	info.play.eof = 0;
138	info.play.error = 0;
139
140	if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
141	{
142	perror("AUDIO_SETINFO");
143	close(g_dsp_fd);
144	return False;
145	}
146
147	return True;
148	}
149
150	void
151	wave_out_write(STREAM s, uint16 tick, uint8 index)
152	{
153	struct audio_packet *packet = &packet_queue[queue_hi];
154	unsigned int next_hi = (queue_hi + 1) % MAX_QUEUE;
155
156	if (next_hi == queue_lo)
157	{
158	error("No space to queue audio packet\n");
159	return;
160	}
161
162	queue_hi = next_hi;
163
164	packet->s = *s;
165	packet->tick = tick;
166	packet->index = index;
167
168	/* we steal the data buffer from s, give it a new one */
169	s->data = malloc(s->size);
170
171	if (!g_dsp_busy)
172	wave_out_play();
173	}
174
175	void
176	wave_out_play(void)
177	{
178	struct audio_packet *packet;
179	audio_info_t info;
180	ssize_t len;
181	unsigned int i;
182	uint8 swap;
183	STREAM out;
184	static BOOL swapped = False;
185	static BOOL sentcompletion = True;
186	static uint32 samplecnt = 0;
187	static uint32 numsamples;
188
189	while (1)
190	{
191	if ( reopened )
192	{
193	/* Device was just (re)openend */
194	samplecnt = 0;
195	swapped = False;
196	sentcompletion = True;
197	reopened = False;
198	}
199
200	if (queue_lo == queue_hi)
201	{
202	g_dsp_busy = 0;
203	return;
204	}
205
206	packet = &packet_queue[queue_lo];
207	out = &packet->s;
208
209	/* Swap the current packet, but only once */
210	if ( swapaudio && ! swapped )
211	{
212	for ( i = 0; i < out->end - out->p; i+=2 )
213	{
214	swap = *(out->p + i);
215	(out->p + i ) = (out->p + i + 1);
216	*(out->p + i + 1) = swap;
217	swapped = True;
218	}
219	}
220
221	if ( sentcompletion )
222	{
223	sentcompletion = False;
224	numsamples = (out->end - out->p)/samplewidth;
225	}
226
227	len=0;
228
229	if ( out->end != out->p )
230	{
231	len = write(g_dsp_fd, out->p, out->end - out->p);
232	if (len == -1)
233	{
234	if (errno != EWOULDBLOCK)
235	perror("write audio");
236	g_dsp_busy = 1;
237	return;
238	}
239	}
240
241	out->p += len;
242	if (out->p == out->end)
243	{
244	if (ioctl(g_dsp_fd, AUDIO_GETINFO, &info) == -1)
245	{
246	perror("AUDIO_GETINFO");
247	return;
248	}
249
250	/* Ack the packet, if we have played at least 70% */
251	if ( info.play.samples >= samplecnt+((numsamples*7)/10) )
252	{
253	samplecnt += numsamples;
254	rdpsnd_send_completion(packet->tick, packet->index);
255	free(out->data);
256	queue_lo = (queue_lo + 1) % MAX_QUEUE;
257	swapped = False;
258	sentcompletion = True;
259	}
260	else
261	{
262	g_dsp_busy = 1;
263	return;
264	}
265	}
266	}
267	}