trunk/rdesktop/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	/* -- 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	}