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

        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;
                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;
        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 ( 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	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			BOOL g_dsp_busy;
35	matthewc	477	static BOOL reopened;
36	matthewc	476	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	matthewc	477	char *dsp_dev = getenv("AUDIODEV");
50	matthewc	476
51	matthewc	477	if ( dsp_dev == NULL )
52			{
53			dsp_dev="/dev/audio";
54			}
55
56	matthewc	476	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	matthewc	477
65	matthewc	476	queue_lo = queue_hi = 0;
66	matthewc	477	reopened = True;
67
68	matthewc	476	return True;
69			}
70
71			void
72			wave_out_close(void)
73			{
74	matthewc	477	/* 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	matthewc	476	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			}
115			else if (pwfx->wBitsPerSample == 16)
116			{
117			info.play.encoding = AUDIO_ENCODING_LINEAR;
118	matthewc	477	/* Do we need to swap the 16bit values? (Are we BigEndian) */
119	matthewc	476	swapaudio = !((uint8 ) (&test));
120			}
121
122	stargo	491	samplewidth = pwfx->wBitsPerSample/8;
123
124	matthewc	476	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	matthewc	477	samplewidth *= 2;
132	matthewc	476	}
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	stargo	491	reopened = True;
140	matthewc	476
141			if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
142			{
143			perror("AUDIO_SETINFO");
144			close(g_dsp_fd);
145			return False;
146			}
147
148			return True;
149			}
150
151			void
152	stargo	491	wave_out_volume(uint16 left, uint16 right)
153			{
154			audio_info_t info;
155			uint balance;
156			uint volume;
157
158			if (ioctl(g_dsp_fd, AUDIO_GETINFO, &info) == -1)
159			{
160			perror("AUDIO_GETINFO");
161			return;
162			}
163
164			volume = (left > right) ? left : right;
165
166			if ( volume/AUDIO_MID_BALANCE != 0 )
167			{
168			balance = AUDIO_MID_BALANCE - (left/(volume/AUDIO_MID_BALANCE)) + (right/(volume/AUDIO_MID_BALANCE));
169			}
170			else
171			{
172			balance = AUDIO_MID_BALANCE;
173			}
174
175			info.play.gain = volume/(65536/AUDIO_MAX_GAIN);
176			info.play.balance = balance;
177
178			if (ioctl(g_dsp_fd, AUDIO_SETINFO, &info) == -1)
179			{
180			perror("AUDIO_SETINFO");
181			return;
182			}
183			}
184
185			void
186	matthewc	476	wave_out_write(STREAM s, uint16 tick, uint8 index)
187			{
188			struct audio_packet *packet = &packet_queue[queue_hi];
189			unsigned int next_hi = (queue_hi + 1) % MAX_QUEUE;
190
191			if (next_hi == queue_lo)
192			{
193			error("No space to queue audio packet\n");
194			return;
195			}
196
197			queue_hi = next_hi;
198
199			packet->s = *s;
200			packet->tick = tick;
201			packet->index = index;
202	stargo	491	packet->s.p += 4;
203	matthewc	476
204			/* we steal the data buffer from s, give it a new one */
205			s->data = malloc(s->size);
206
207			if (!g_dsp_busy)
208			wave_out_play();
209			}
210
211			void
212			wave_out_play(void)
213			{
214			struct audio_packet *packet;
215			audio_info_t info;
216			ssize_t len;
217			unsigned int i;
218			uint8 swap;
219			STREAM out;
220			static BOOL swapped = False;
221			static BOOL sentcompletion = True;
222	matthewc	477	static uint32 samplecnt = 0;
223			static uint32 numsamples;
224	matthewc	476
225			while (1)
226			{
227	matthewc	477	if ( reopened )
228			{
229			/* Device was just (re)openend */
230			samplecnt = 0;
231			swapped = False;
232			sentcompletion = True;
233			reopened = False;
234			}
235
236	matthewc	476	if (queue_lo == queue_hi)
237			{
238			g_dsp_busy = 0;
239			return;
240			}
241
242			packet = &packet_queue[queue_lo];
243			out = &packet->s;
244
245	matthewc	477	/* Swap the current packet, but only once */
246	matthewc	476	if ( swapaudio && ! swapped )
247			{
248	stargo	491	for ( i = 0; i < out->end - out->p; i += 2 )
249	matthewc	476	{
250			swap = *(out->p + i);
251	stargo	491	(out->p + i) = (out->p + i + 1);
252	matthewc	476	*(out->p + i + 1) = swap;
253			}
254	stargo	491	swapped = True;
255	matthewc	476	}
256
257			if ( sentcompletion )
258			{
259			sentcompletion = False;
260	matthewc	477	numsamples = (out->end - out->p)/samplewidth;
261	matthewc	476	}
262
263			len=0;
264
265	matthewc	477	if ( out->end != out->p )
266	matthewc	476	{
267			len = write(g_dsp_fd, out->p, out->end - out->p);
268			if (len == -1)
269			{
270			if (errno != EWOULDBLOCK)
271			perror("write audio");
272			g_dsp_busy = 1;
273			return;
274			}
275			}
276
277			out->p += len;
278			if (out->p == out->end)
279			{
280			if (ioctl(g_dsp_fd, AUDIO_GETINFO, &info) == -1)
281			{
282			perror("AUDIO_GETINFO");
283			return;
284			}
285	matthewc	477
286			/* Ack the packet, if we have played at least 70% */
287			if ( info.play.samples >= samplecnt+((numsamples*7)/10) )
288	matthewc	476	{
289	matthewc	477	samplecnt += numsamples;
290	matthewc	476	rdpsnd_send_completion(packet->tick, packet->index);
291			free(out->data);
292			queue_lo = (queue_lo + 1) % MAX_QUEUE;
293			swapped = False;
294			sentcompletion = True;
295			}
296			else
297			{
298			g_dsp_busy = 1;
299			return;
300			}
301			}
302			}
303			}