rdesktop/trunk/rdpsnd_alsa.c

/* -*- c-basic-offset: 8 -*-
   rdesktop: A Remote Desktop Protocol client.
   Sound Channel Process Functions - alsa-driver
   Copyright (C) Matthew Chapman 2003
   Copyright (C) GuoJunBo guojunbo@ict.ac.cn 2003
   Copyright (C) Michael Gernoth mike@zerfleddert.de 2006

   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 <alsa/asoundlib.h>
#include <sys/time.h>

#define DEFAULTDEVICE   "default"
#define MAX_QUEUE       10
#define MAX_FRAMES      32

int g_dsp_fd;
BOOL g_dsp_busy = False;
static snd_pcm_t *pcm_handle = NULL;
static snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;
static BOOL reopened;
static short samplewidth;
static int audiochannels;

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 *pcm_name;
        int err;

        pcm_name = xstrdup(DEFAULTDEVICE);

        if ((err = snd_pcm_open(&pcm_handle, pcm_name, stream, 0)) < 0)
        {
                error("snd_pcm_open: %s\n", snd_strerror(err));
                return False;
        }

        g_dsp_fd = 0;
        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;
        }

        if (pcm_handle)
        {
                snd_pcm_drop(pcm_handle);
                snd_pcm_close(pcm_handle);
        }
}

BOOL
wave_out_format_supported(WAVEFORMATEX * pwfx)
{
#if 0
        int err;
        snd_pcm_hw_params_t *hwparams = NULL;

        if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
        {
                error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
                return False;
        }

        if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
        {
                error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
                return False;
        }
        snd_pcm_hw_params_free(hwparams);
#endif

        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;
        if ((pwfx->nSamplesPerSec != 44100) && (pwfx->nSamplesPerSec != 22050))
                return False;

        return True;
}

BOOL
wave_out_set_format(WAVEFORMATEX * pwfx)
{
        snd_pcm_hw_params_t *hwparams = NULL;
        unsigned int rate, exact_rate;
        int err;
        unsigned int buffertime;

        samplewidth = pwfx->wBitsPerSample / 8;

        if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
        {
                error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
                return False;
        }

        if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
        {
                error("snd_pcm_hw_params_any: %s\n", snd_strerror(err));
                return False;
        }

        if ((err =
             snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
        {
                error("snd_pcm_hw_params_set_access: %s\n", snd_strerror(err));
                return False;
        }

        if (pwfx->wBitsPerSample == 16)
        {
                if ((err =
                     snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
                {
                        error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
                        return False;
                }
        }
        else
        {
                if ((err =
                     snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S8)) < 0)
                {
                        error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
                        return False;
                }
        }

#if 0
        if ((err = snd_pcm_hw_params_set_rate_resample(pcm_handle, hwparams, 1)) < 0)
        {
                error("snd_pcm_hw_params_set_rate_resample: %s\n", snd_strerror(err));
                return False;
        }
#endif

        exact_rate = rate = pwfx->nSamplesPerSec;
        if ((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_rate, 0)) < 0)
        {
                error("snd_pcm_hw_params_set_rate_near: %s\n", snd_strerror(err));
                return False;
        }

        audiochannels = pwfx->nChannels;
        if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, pwfx->nChannels)) < 0)
        {
                error("snd_pcm_hw_params_set_channels: %s\n", snd_strerror(err));
                return False;
        }


        buffertime = 500000;    /* microseconds */
        if ((err =
             snd_pcm_hw_params_set_buffer_time_near(pcm_handle, hwparams, &buffertime, 0)) < 0)
        {
                error("snd_pcm_hw_params_set_buffer_time_near: %s\n", snd_strerror(err));
                return False;
        }

        if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0)
        {
                error("snd_pcm_hw_params: %s\n", snd_strerror(err));
                return False;
        }

        snd_pcm_hw_params_free(hwparams);

        if ((err = snd_pcm_prepare(pcm_handle)) < 0)
        {
                error("snd_pcm_prepare: %s\n", snd_strerror(err));
                return False;
        }

        reopened = True;

        return True;
}

void
wave_out_volume(uint16 left, uint16 right)
{
        static int warned = 0;

        if (!warned)
        {
                warning("volume changes currently not supported with experimental alsa-output\n");
                warned = 1;
        }
}

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 = (uint8 *) malloc(s->size);

        if (!g_dsp_busy)
                wave_out_play();
}

void
wave_out_play(void)
{
        struct audio_packet *packet;
        STREAM out;
        int len;
        static long prev_s, prev_us;
        unsigned int duration;
        struct timeval tv;
        int next_tick;

        if (reopened)
        {
                reopened = False;
                gettimeofday(&tv, NULL);
                prev_s = tv.tv_sec;
                prev_us = tv.tv_usec;
        }

        if (queue_lo == queue_hi)
        {
                g_dsp_busy = 0;
                return;
        }

        packet = &packet_queue[queue_lo];
        out = &packet->s;

        if (((queue_lo + 1) % MAX_QUEUE) != queue_hi)
        {
                next_tick = packet_queue[(queue_lo + 1) % MAX_QUEUE].tick;
        }
        else
        {
                next_tick = (packet->tick + 65535) % 65536;
        }

        len = (out->end - out->p) / (samplewidth * audiochannels);
        if ((len = snd_pcm_writei(pcm_handle, out->p, ((MAX_FRAMES < len) ? MAX_FRAMES : len))) < 0)
        {
                snd_pcm_prepare(pcm_handle);
                len = 0;
        }
        out->p += (len * samplewidth * audiochannels);

        gettimeofday(&tv, NULL);

        duration = ((tv.tv_sec - prev_s) * 1000000 + (tv.tv_usec - prev_us)) / 1000;

        if (packet->tick > next_tick)
                next_tick += 65536;

        if ((out->p == out->end) || duration > next_tick - packet->tick + 500)
        {
                prev_s = tv.tv_sec;
                prev_us = tv.tv_usec;

                if (abs((next_tick - packet->tick) - duration) > 20)
                {
                        DEBUG(("duration: %d, calc: %d, ", duration, next_tick - packet->tick));
                        DEBUG(("last: %d, is: %d, should: %d\n", packet->tick,
                               (packet->tick + duration) % 65536, next_tick % 65536));
                }

                /* Until all drivers are using the windows sound-ticks, we need to
                   substract the 50 ticks added later by rdpsnd.c */
                rdpsnd_send_completion(((packet->tick + duration) % 65536) - 50, packet->index);
                free(out->data);
                queue_lo = (queue_lo + 1) % MAX_QUEUE;
        }

        g_dsp_busy = 1;
        return;
}
1	stargo	1253	/* -- c-basic-offset: 8 --
2			rdesktop: A Remote Desktop Protocol client.
3			Sound Channel Process Functions - alsa-driver
4			Copyright (C) Matthew Chapman 2003
5			Copyright (C) GuoJunBo guojunbo@ict.ac.cn 2003
6			Copyright (C) Michael Gernoth mike@zerfleddert.de 2006
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 <alsa/asoundlib.h>
28			#include <sys/time.h>
29
30			#define DEFAULTDEVICE "default"
31			#define MAX_QUEUE 10
32			#define MAX_FRAMES 32
33
34			int g_dsp_fd;
35			BOOL g_dsp_busy = False;
36			static snd_pcm_t *pcm_handle = NULL;
37			static snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;
38			static BOOL reopened;
39			static short samplewidth;
40			static int audiochannels;
41
42			static struct audio_packet
43			{
44			struct stream s;
45			uint16 tick;
46			uint8 index;
47			} packet_queue[MAX_QUEUE];
48			static unsigned int queue_hi, queue_lo;
49
50			BOOL
51			wave_out_open(void)
52			{
53			char *pcm_name;
54			int err;
55
56			pcm_name = xstrdup(DEFAULTDEVICE);
57
58			if ((err = snd_pcm_open(&pcm_handle, pcm_name, stream, 0)) < 0)
59			{
60			error("snd_pcm_open: %s\n", snd_strerror(err));
61			return False;
62			}
63
64			g_dsp_fd = 0;
65			queue_lo = queue_hi = 0;
66
67			reopened = True;
68
69			return True;
70			}
71
72			void
73			wave_out_close(void)
74			{
75			/* Ack all remaining packets */
76			while (queue_lo != queue_hi)
77			{
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			if (pcm_handle)
84			{
85			snd_pcm_drop(pcm_handle);
86			snd_pcm_close(pcm_handle);
87			}
88			}
89
90			BOOL
91			wave_out_format_supported(WAVEFORMATEX * pwfx)
92			{
93			#if 0
94			int err;
95			snd_pcm_hw_params_t *hwparams = NULL;
96
97			if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
98			{
99			error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
100			return False;
101			}
102
103			if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
104			{
105			error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
106			return False;
107			}
108			snd_pcm_hw_params_free(hwparams);
109			#endif
110
111			if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
112			return False;
113			if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
114			return False;
115			if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
116			return False;
117			if ((pwfx->nSamplesPerSec != 44100) && (pwfx->nSamplesPerSec != 22050))
118			return False;
119
120			return True;
121			}
122
123			BOOL
124			wave_out_set_format(WAVEFORMATEX * pwfx)
125			{
126			snd_pcm_hw_params_t *hwparams = NULL;
127			unsigned int rate, exact_rate;
128			int err;
129			unsigned int buffertime;
130
131			samplewidth = pwfx->wBitsPerSample / 8;
132
133			if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
134			{
135			error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
136			return False;
137			}
138
139			if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
140			{
141			error("snd_pcm_hw_params_any: %s\n", snd_strerror(err));
142			return False;
143			}
144
145			if ((err =
146			snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
147			{
148			error("snd_pcm_hw_params_set_access: %s\n", snd_strerror(err));
149			return False;
150			}
151
152			if (pwfx->wBitsPerSample == 16)
153			{
154			if ((err =
155			snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
156			{
157			error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
158			return False;
159			}
160			}
161			else
162			{
163			if ((err =
164			snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S8)) < 0)
165			{
166			error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
167			return False;
168			}
169			}
170
171			#if 0
172			if ((err = snd_pcm_hw_params_set_rate_resample(pcm_handle, hwparams, 1)) < 0)
173			{
174			error("snd_pcm_hw_params_set_rate_resample: %s\n", snd_strerror(err));
175			return False;
176			}
177			#endif
178
179			exact_rate = rate = pwfx->nSamplesPerSec;
180			if ((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_rate, 0)) < 0)
181			{
182			error("snd_pcm_hw_params_set_rate_near: %s\n", snd_strerror(err));
183			return False;
184			}
185
186			audiochannels = pwfx->nChannels;
187			if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, pwfx->nChannels)) < 0)
188			{
189			error("snd_pcm_hw_params_set_channels: %s\n", snd_strerror(err));
190			return False;
191			}
192
193
194			buffertime = 500000; /* microseconds */
195			if ((err =
196			snd_pcm_hw_params_set_buffer_time_near(pcm_handle, hwparams, &buffertime, 0)) < 0)
197			{
198			error("snd_pcm_hw_params_set_buffer_time_near: %s\n", snd_strerror(err));
199			return False;
200			}
201
202			if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0)
203			{
204			error("snd_pcm_hw_params: %s\n", snd_strerror(err));
205			return False;
206			}
207
208			snd_pcm_hw_params_free(hwparams);
209
210			if ((err = snd_pcm_prepare(pcm_handle)) < 0)
211			{
212			error("snd_pcm_prepare: %s\n", snd_strerror(err));
213			return False;
214			}
215
216			reopened = True;
217
218			return True;
219			}
220
221			void
222			wave_out_volume(uint16 left, uint16 right)
223			{
224			static int warned = 0;
225
226			if (!warned)
227			{
228			warning("volume changes currently not supported with experimental alsa-output\n");
229			warned = 1;
230			}
231			}
232
233			void
234			wave_out_write(STREAM s, uint16 tick, uint8 index)
235			{
236			struct audio_packet *packet = &packet_queue[queue_hi];
237			unsigned int next_hi = (queue_hi + 1) % MAX_QUEUE;
238
239			if (next_hi == queue_lo)
240			{
241			error("No space to queue audio packet\n");
242			return;
243			}
244
245			queue_hi = next_hi;
246
247			packet->s = *s;
248			packet->tick = tick;
249			packet->index = index;
250			packet->s.p += 4;
251
252			/* we steal the data buffer from s, give it a new one */
253			s->data = (uint8 *) malloc(s->size);
254
255			if (!g_dsp_busy)
256			wave_out_play();
257			}
258
259			void
260			wave_out_play(void)
261			{
262			struct audio_packet *packet;
263			STREAM out;
264			int len;
265			static long prev_s, prev_us;
266			unsigned int duration;
267			struct timeval tv;
268			int next_tick;
269
270			if (reopened)
271			{
272			reopened = False;
273			gettimeofday(&tv, NULL);
274			prev_s = tv.tv_sec;
275			prev_us = tv.tv_usec;
276			}
277
278			if (queue_lo == queue_hi)
279			{
280			g_dsp_busy = 0;
281			return;
282			}
283
284			packet = &packet_queue[queue_lo];
285			out = &packet->s;
286
287			if (((queue_lo + 1) % MAX_QUEUE) != queue_hi)
288			{
289			next_tick = packet_queue[(queue_lo + 1) % MAX_QUEUE].tick;
290			}
291			else
292			{
293			next_tick = (packet->tick + 65535) % 65536;
294			}
295
296			len = (out->end - out->p) / (samplewidth * audiochannels);
297			if ((len = snd_pcm_writei(pcm_handle, out->p, ((MAX_FRAMES < len) ? MAX_FRAMES : len))) < 0)
298			{
299			snd_pcm_prepare(pcm_handle);
300			len = 0;
301			}
302			out->p += (len * samplewidth * audiochannels);
303
304			gettimeofday(&tv, NULL);
305
306			duration = ((tv.tv_sec - prev_s) * 1000000 + (tv.tv_usec - prev_us)) / 1000;
307
308			if (packet->tick > next_tick)
309			next_tick += 65536;
310
311			if ((out->p == out->end) \|\| duration > next_tick - packet->tick + 500)
312			{
313			prev_s = tv.tv_sec;
314			prev_us = tv.tv_usec;
315
316			if (abs((next_tick - packet->tick) - duration) > 20)
317			{
318			DEBUG(("duration: %d, calc: %d, ", duration, next_tick - packet->tick));
319			DEBUG(("last: %d, is: %d, should: %d\n", packet->tick,
320			(packet->tick + duration) % 65536, next_tick % 65536));
321			}
322
323			/* Until all drivers are using the windows sound-ticks, we need to
324			substract the 50 ticks added later by rdpsnd.c */
325			rdpsnd_send_completion(((packet->tick + duration) % 65536) - 50, packet->index);
326			free(out->data);
327			queue_lo = (queue_lo + 1) % MAX_QUEUE;
328			}
329
330			g_dsp_busy = 1;
331			return;
332			}