rdesktop/trunk/rdpsnd_sgi.c

/* 
   rdesktop: A Remote Desktop Protocol client.
   Sound Channel Process Functions - SGI/IRIX
   Copyright (C) Matthew Chapman 2003
   Copyright (C) GuoJunBo guojunbo@ict.ac.cn 2003
   Copyright (C) Jeremy Meng voidfoo@cwazy.co.uk 2004

   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 <errno.h>
#include <dmedia/audio.h>

#define IRIX_DEBUG 1

#define IRIX_MAX_VOL     65535

#define MAX_QUEUE       10

int g_dsp_fd;
ALconfig audioconfig;
ALport output_port;

BOOL g_dsp_busy = False;
static BOOL g_swapaudio;
static int g_snd_rate;
static BOOL g_swapaudio;
static short g_samplewidth;
static int width = AL_SAMPLE_16;

double min_volume, max_volume, volume_range;
int resource, maxFillable;
int combinedFrameSize;

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)
{
        ALparamInfo pinfo;

#if (defined(IRIX_DEBUG))
        fprintf(stderr, "wave_out_open: begin\n");
#endif

        if (alGetParamInfo(AL_DEFAULT_OUTPUT, AL_GAIN, &pinfo) < 0)
        {
                fprintf(stderr, "wave_out_open: alGetParamInfo failed: %s\n",
                        alGetErrorString(oserror()));
        }
        min_volume = alFixedToDouble(pinfo.min.ll);
        max_volume = alFixedToDouble(pinfo.max.ll);
        volume_range = (max_volume - min_volume);
#if (defined(IRIX_DEBUG))
        fprintf(stderr, "wave_out_open: minvol = %lf, maxvol= %lf, range = %lf.\n",
                min_volume, max_volume, volume_range);
#endif

        queue_lo = queue_hi = 0;

        audioconfig = alNewConfig();
        if (audioconfig < 0)
        {
                fprintf(stderr, "wave_out_open: alNewConfig failed: %s\n",
                        alGetErrorString(oserror()));
                return False;
        }

        output_port = alOpenPort("rdpsnd", "w", 0);
        if (output_port == (ALport) 0)
        {
                fprintf(stderr, "wave_out_open: alOpenPort failed: %s\n",
                        alGetErrorString(oserror()));
                return False;
        }

#if (defined(IRIX_DEBUG))
        fprintf(stderr, "wave_out_open: returning\n");
#endif
        return True;
}

void
wave_out_close(void)
{
        /* Ack all remaining packets */
#if (defined(IRIX_DEBUG))
        fprintf(stderr, "wave_out_close: begin\n");
#endif

        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;
        }
        alDiscardFrames(output_port, 0);

        alClosePort(output_port);
        alFreeConfig(audioconfig);
#if (defined(IRIX_DEBUG))
        fprintf(stderr, "wave_out_close: returning\n");
#endif
}

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)
{
        int channels;
        int frameSize, channelCount;
        ALpv params;

#if (defined(IRIX_DEBUG))
        fprintf(stderr, "wave_out_set_format: init...\n");
#endif
        /* limited support to configure an opened audio port in IRIX */
        /* have to reopen the audio port, using same config */
        alClosePort(output_port);

        g_swapaudio = False;

        if (pwfx->wBitsPerSample == 8)
                width = AL_SAMPLE_8;
        else if (pwfx->wBitsPerSample == 16)
        {
                width = AL_SAMPLE_16;
                /* Do we need to swap the 16bit values? (Are we BigEndian) */
#if (defined(IRIX_DEBUG))
                g_swapaudio = 1;
#else
                g_swapaudio = 0;
#endif
        }

        g_samplewidth = pwfx->wBitsPerSample / 8;
        channels = pwfx->nChannels;
        g_snd_rate = pwfx->nSamplesPerSec;

        alSetSampFmt(audioconfig, AL_SAMPFMT_TWOSCOMP);
        alSetWidth(audioconfig, width);
        alSetChannels(audioconfig, channels);

        output_port = alOpenPort("rdpsnd", "w", audioconfig);

        if (output_port == (ALport) 0)
        {
                fprintf(stderr, "wave_out_set_format: alOpenPort failed: %s\n",
                        alGetErrorString(oserror()));
                return False;
        }

        resource = alGetResource(output_port);
        maxFillable = alGetFillable(output_port);
        channelCount = alGetChannels(audioconfig);
        frameSize = alGetWidth(audioconfig);

        if (frameSize == 0 || channelCount == 0)
        {
                fprintf(stderr, "wave_out_set_format: bad frameSize or channelCount\n");
                return False;
        }
        combinedFrameSize = frameSize * channelCount;

        params.param = AL_RATE;
        params.value.ll = (long long) g_snd_rate << 32;

        if (alSetParams(resource, &params, 1) < 0)
        {
                fprintf(stderr, "wave_set_format: alSetParams failed: %s\n",
                        alGetErrorString(oserror()));
                return False;
        }
        if (params.sizeOut < 0)
        {
                fprintf(stderr, "wave_set_format: invalid rate %d\n", g_snd_rate);
                return False;
        }

#if (defined(IRIX_DEBUG))
        fprintf(stderr, "wave_out_set_format: returning...\n");
#endif
        return True;
}

void
wave_out_volume(uint16 left, uint16 right)
{
        double gainleft, gainright;
        ALpv pv[1];
        ALfixed gain[8];

#if (defined(IRIX_DEBUG))
        fprintf(stderr, "wave_out_volume: begin\n");
        fprintf(stderr, "left='%d', right='%d'\n", left, right);
#endif

        gainleft = (double) left / IRIX_MAX_VOL;
        gainright = (double) right / IRIX_MAX_VOL;

        gain[0] = alDoubleToFixed(min_volume + gainleft * volume_range);
        gain[1] = alDoubleToFixed(min_volume + gainright * volume_range);

        pv[0].param = AL_GAIN;
        pv[0].value.ptr = gain;
        pv[0].sizeIn = 8;
        if (alSetParams(AL_DEFAULT_OUTPUT, pv, 1) < 0)
        {
                fprintf(stderr, "wave_out_volume: alSetParams failed: %s\n",
                        alGetErrorString(oserror()));
                return;
        }

#if (defined(IRIX_DEBUG))
        fprintf(stderr, "wave_out_volume: returning\n");
#endif
}

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)
        {
                fprintf(stderr, "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;
        ssize_t len;
        unsigned int i;
        uint8 swap;
        STREAM out;
        static long startedat_us;
        static long startedat_s;
        static BOOL started = False;
        static BOOL swapped = False;
        struct timeval tv;
        int gf;
        static long long temp;

        while (1)
        {
                if (queue_lo == queue_hi)
                {
                        g_dsp_busy = False;
                        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 (!started)
                {
                        gettimeofday(&tv, NULL);
                        startedat_us = tv.tv_usec;
                        startedat_s = tv.tv_sec;
                        started = True;
                }

                len = out->end - out->p;
                gf = alGetFillable(output_port);
                if (len > gf)
                {
                        //len = gf * combinedFrameSize;
#if (defined(IRIX_DEBUG))
                        //fprintf(stderr,"Fillable...\n");
#endif
                }

                alWriteFrames(output_port, out->p, len / combinedFrameSize);

                out->p += len;
                if (out->p == out->end)
                {
                        long long duration;
                        long elapsed;

                        gettimeofday(&tv, NULL);
                        duration = (out->size * (1000000 / (g_samplewidth * g_snd_rate)));
                        elapsed = (tv.tv_sec - startedat_s) * 1000000 + (tv.tv_usec - startedat_us);
                        /* 7/10 is not good for IRIX audio port, 4x/100 is suitable */
                        if (elapsed >= (duration * 485) / 1000)
                        {
                                rdpsnd_send_completion(packet->tick, packet->index);
                                free(out->data);
                                queue_lo = (queue_lo + 1) % MAX_QUEUE;
                                started = False;
                                swapped = False;
                        }
                        else
                        {
#if (defined(IRIX_DEBUG))
                                //fprintf(stderr,"Busy playing...\n");
#endif
                                g_dsp_busy = True;
                                return;
                        }
                }
        }
}
1	stargo	744	/*
2			rdesktop: A Remote Desktop Protocol client.
3			Sound Channel Process Functions - SGI/IRIX
4			Copyright (C) Matthew Chapman 2003
5			Copyright (C) GuoJunBo guojunbo@ict.ac.cn 2003
6	stargo	784	Copyright (C) Jeremy Meng voidfoo@cwazy.co.uk 2004
7	stargo	744
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 <errno.h>
25			#include <dmedia/audio.h>
26
27			#define IRIX_DEBUG 1
28
29			#define IRIX_MAX_VOL 65535
30
31			#define MAX_QUEUE 10
32
33			int g_dsp_fd;
34			ALconfig audioconfig;
35			ALport output_port;
36
37			BOOL g_dsp_busy = False;
38			static BOOL g_swapaudio;
39			static int g_snd_rate;
40			static BOOL g_swapaudio;
41			static short g_samplewidth;
42			static int width = AL_SAMPLE_16;
43
44			double min_volume, max_volume, volume_range;
45			int resource, maxFillable;
46			int combinedFrameSize;
47
48			static struct audio_packet
49			{
50			struct stream s;
51			uint16 tick;
52			uint8 index;
53			} packet_queue[MAX_QUEUE];
54			static unsigned int queue_hi, queue_lo;
55
56			BOOL
57			wave_out_open(void)
58			{
59	astrand	746	ALparamInfo pinfo;
60	stargo	744
61			#if (defined(IRIX_DEBUG))
62	astrand	746	fprintf(stderr, "wave_out_open: begin\n");
63	stargo	744	#endif
64
65	astrand	746	if (alGetParamInfo(AL_DEFAULT_OUTPUT, AL_GAIN, &pinfo) < 0)
66			{
67			fprintf(stderr, "wave_out_open: alGetParamInfo failed: %s\n",
68			alGetErrorString(oserror()));
69			}
70			min_volume = alFixedToDouble(pinfo.min.ll);
71			max_volume = alFixedToDouble(pinfo.max.ll);
72			volume_range = (max_volume - min_volume);
73	stargo	744	#if (defined(IRIX_DEBUG))
74	astrand	746	fprintf(stderr, "wave_out_open: minvol = %lf, maxvol= %lf, range = %lf.\n",
75			min_volume, max_volume, volume_range);
76	stargo	744	#endif
77
78			queue_lo = queue_hi = 0;
79
80			audioconfig = alNewConfig();
81	astrand	746	if (audioconfig < 0)
82			{
83			fprintf(stderr, "wave_out_open: alNewConfig failed: %s\n",
84			alGetErrorString(oserror()));
85	stargo	744	return False;
86			}
87
88			output_port = alOpenPort("rdpsnd", "w", 0);
89	astrand	746	if (output_port == (ALport) 0)
90			{
91			fprintf(stderr, "wave_out_open: alOpenPort failed: %s\n",
92			alGetErrorString(oserror()));
93	stargo	744	return False;
94			}
95
96			#if (defined(IRIX_DEBUG))
97	astrand	746	fprintf(stderr, "wave_out_open: returning\n");
98	stargo	744	#endif
99			return True;
100			}
101
102			void
103			wave_out_close(void)
104			{
105			/* Ack all remaining packets */
106			#if (defined(IRIX_DEBUG))
107	astrand	746	fprintf(stderr, "wave_out_close: begin\n");
108	stargo	744	#endif
109	astrand	746
110	stargo	744	while (queue_lo != queue_hi)
111			{
112			rdpsnd_send_completion(packet_queue[queue_lo].tick, packet_queue[queue_lo].index);
113			free(packet_queue[queue_lo].s.data);
114			queue_lo = (queue_lo + 1) % MAX_QUEUE;
115			}
116	astrand	746	alDiscardFrames(output_port, 0);
117
118	stargo	744	alClosePort(output_port);
119			alFreeConfig(audioconfig);
120			#if (defined(IRIX_DEBUG))
121	astrand	746	fprintf(stderr, "wave_out_close: returning\n");
122	stargo	744	#endif
123			}
124
125			BOOL
126			wave_out_format_supported(WAVEFORMATEX * pwfx)
127			{
128			if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
129			return False;
130			if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
131			return False;
132			if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
133			return False;
134
135			return True;
136			}
137
138			BOOL
139			wave_out_set_format(WAVEFORMATEX * pwfx)
140			{
141			int channels;
142	astrand	746	int frameSize, channelCount;
143			ALpv params;
144
145	stargo	744	#if (defined(IRIX_DEBUG))
146	astrand	746	fprintf(stderr, "wave_out_set_format: init...\n");
147	stargo	744	#endif
148			/* limited support to configure an opened audio port in IRIX */
149			/* have to reopen the audio port, using same config */
150			alClosePort(output_port);
151
152			g_swapaudio = False;
153
154			if (pwfx->wBitsPerSample == 8)
155			width = AL_SAMPLE_8;
156	astrand	746	else if (pwfx->wBitsPerSample == 16)
157			{
158	stargo	744	width = AL_SAMPLE_16;
159			/* Do we need to swap the 16bit values? (Are we BigEndian) */
160			#if (defined(IRIX_DEBUG))
161			g_swapaudio = 1;
162			#else
163			g_swapaudio = 0;
164			#endif
165			}
166
167			g_samplewidth = pwfx->wBitsPerSample / 8;
168			channels = pwfx->nChannels;
169			g_snd_rate = pwfx->nSamplesPerSec;
170
171			alSetSampFmt(audioconfig, AL_SAMPFMT_TWOSCOMP);
172			alSetWidth(audioconfig, width);
173			alSetChannels(audioconfig, channels);
174
175			output_port = alOpenPort("rdpsnd", "w", audioconfig);
176
177	astrand	746	if (output_port == (ALport) 0)
178			{
179			fprintf(stderr, "wave_out_set_format: alOpenPort failed: %s\n",
180			alGetErrorString(oserror()));
181	stargo	744	return False;
182			}
183
184			resource = alGetResource(output_port);
185			maxFillable = alGetFillable(output_port);
186	astrand	746	channelCount = alGetChannels(audioconfig);
187			frameSize = alGetWidth(audioconfig);
188	stargo	744
189	astrand	746	if (frameSize == 0 \|\| channelCount == 0)
190			{
191			fprintf(stderr, "wave_out_set_format: bad frameSize or channelCount\n");
192			return False;
193			}
194			combinedFrameSize = frameSize * channelCount;
195	stargo	744
196	astrand	746	params.param = AL_RATE;
197			params.value.ll = (long long) g_snd_rate << 32;
198	stargo	744
199	astrand	746	if (alSetParams(resource, &params, 1) < 0)
200			{
201			fprintf(stderr, "wave_set_format: alSetParams failed: %s\n",
202			alGetErrorString(oserror()));
203			return False;
204			}
205			if (params.sizeOut < 0)
206			{
207			fprintf(stderr, "wave_set_format: invalid rate %d\n", g_snd_rate);
208			return False;
209			}
210	stargo	744
211			#if (defined(IRIX_DEBUG))
212	astrand	746	fprintf(stderr, "wave_out_set_format: returning...\n");
213	stargo	744	#endif
214			return True;
215			}
216
217			void
218			wave_out_volume(uint16 left, uint16 right)
219			{
220	astrand	746	double gainleft, gainright;
221			ALpv pv[1];
222			ALfixed gain[8];
223	stargo	744
224			#if (defined(IRIX_DEBUG))
225	astrand	746	fprintf(stderr, "wave_out_volume: begin\n");
226			fprintf(stderr, "left='%d', right='%d'\n", left, right);
227	stargo	744	#endif
228	astrand	746
229			gainleft = (double) left / IRIX_MAX_VOL;
230	stargo	744	gainright = (double) right / IRIX_MAX_VOL;
231
232	astrand	746	gain[0] = alDoubleToFixed(min_volume + gainleft * volume_range);
233			gain[1] = alDoubleToFixed(min_volume + gainright * volume_range);
234	stargo	744
235	astrand	746	pv[0].param = AL_GAIN;
236			pv[0].value.ptr = gain;
237			pv[0].sizeIn = 8;
238			if (alSetParams(AL_DEFAULT_OUTPUT, pv, 1) < 0)
239			{
240			fprintf(stderr, "wave_out_volume: alSetParams failed: %s\n",
241			alGetErrorString(oserror()));
242			return;
243			}
244	stargo	744
245			#if (defined(IRIX_DEBUG))
246	astrand	746	fprintf(stderr, "wave_out_volume: returning\n");
247	stargo	744	#endif
248			}
249
250			void
251			wave_out_write(STREAM s, uint16 tick, uint8 index)
252			{
253			struct audio_packet *packet = &packet_queue[queue_hi];
254			unsigned int next_hi = (queue_hi + 1) % MAX_QUEUE;
255
256			if (next_hi == queue_lo)
257			{
258			fprintf(stderr, "No space to queue audio packet\n");
259			return;
260			}
261
262			queue_hi = next_hi;
263
264			packet->s = *s;
265			packet->tick = tick;
266			packet->index = index;
267			packet->s.p += 4;
268
269			/* we steal the data buffer from s, give it a new one */
270			s->data = malloc(s->size);
271
272			if (!g_dsp_busy)
273			wave_out_play();
274			}
275
276			void
277			wave_out_play(void)
278			{
279			struct audio_packet *packet;
280			ssize_t len;
281			unsigned int i;
282			uint8 swap;
283			STREAM out;
284			static long startedat_us;
285			static long startedat_s;
286			static BOOL started = False;
287			static BOOL swapped = False;
288			struct timeval tv;
289			int gf;
290			static long long temp;
291
292			while (1)
293			{
294			if (queue_lo == queue_hi)
295			{
296			g_dsp_busy = False;
297			return;
298			}
299
300			packet = &packet_queue[queue_lo];
301			out = &packet->s;
302
303			/* Swap the current packet, but only once */
304			if (g_swapaudio && !swapped)
305			{
306			for (i = 0; i < out->end - out->p; i += 2)
307			{
308			swap = *(out->p + i);
309			(out->p + i) = (out->p + i + 1);
310			*(out->p + i + 1) = swap;
311			}
312			swapped = True;
313			}
314
315			if (!started)
316			{
317			gettimeofday(&tv, NULL);
318			startedat_us = tv.tv_usec;
319			startedat_s = tv.tv_sec;
320			started = True;
321			}
322
323			len = out->end - out->p;
324	astrand	746	gf = alGetFillable(output_port);
325			if (len > gf)
326			{
327	stargo	744	//len = gf * combinedFrameSize;
328			#if (defined(IRIX_DEBUG))
329			//fprintf(stderr,"Fillable...\n");
330			#endif
331			}
332
333	astrand	746	alWriteFrames(output_port, out->p, len / combinedFrameSize);
334	stargo	744
335			out->p += len;
336			if (out->p == out->end)
337			{
338			long long duration;
339			long elapsed;
340
341			gettimeofday(&tv, NULL);
342			duration = (out->size * (1000000 / (g_samplewidth * g_snd_rate)));
343	astrand	746	elapsed = (tv.tv_sec - startedat_s) * 1000000 + (tv.tv_usec - startedat_us);
344	stargo	744	/* 7/10 is not good for IRIX audio port, 4x/100 is suitable */
345			if (elapsed >= (duration * 485) / 1000)
346			{
347			rdpsnd_send_completion(packet->tick, packet->index);
348			free(out->data);
349			queue_lo = (queue_lo + 1) % MAX_QUEUE;
350			started = False;
351			swapped = False;
352			}
353			else
354			{
355			#if (defined(IRIX_DEBUG))
356			//fprintf(stderr,"Busy playing...\n");
357			#endif
358			g_dsp_busy = True;
359			return;
360			}
361			}
362			}
363			}