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 "rdpsnd.h"
#include <unistd.h>
#include <fcntl.h>
#include <errno.h>
#include <alsa/asoundlib.h>
#include <sys/time.h>

#define DEFAULTDEVICE   "default"
#define MAX_FRAMES      32

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 char *pcm_name;

BOOL
alsa_open(void)
{
        int err;

        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;
        rdpsnd_queue_init();

        reopened = True;

        return True;
}

void
alsa_close(void)
{
        /* Ack all remaining packets */
        while (!rdpsnd_queue_empty())
        {
                rdpsnd_send_completion(rdpsnd_queue_current_packet()->tick,
                                       rdpsnd_queue_current_packet()->index);
                rdpsnd_queue_next();
        }

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

BOOL
alsa_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
alsa_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
alsa_volume(uint16 left, uint16 right)
{
        static int warned = 0;

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

void
alsa_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 (rdpsnd_queue_empty())
        {
                g_dsp_busy = 0;
                return;
        }

        packet = rdpsnd_queue_current_packet();
        out = &packet->s;

        next_tick = rdpsnd_queue_next_tick();

        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));
                }

                rdpsnd_send_completion(((packet->tick + duration) % 65536), packet->index);
                rdpsnd_queue_next();
        }

        g_dsp_busy = 1;
        return;
}

struct audio_driver *
alsa_register(char *options)
{
        static struct audio_driver alsa_driver;

        alsa_driver.wave_out_write = rdpsnd_queue_write;
        alsa_driver.wave_out_open = alsa_open;
        alsa_driver.wave_out_close = alsa_close;
        alsa_driver.wave_out_format_supported = alsa_format_supported;
        alsa_driver.wave_out_set_format = alsa_set_format;
        alsa_driver.wave_out_volume = alsa_volume;
        alsa_driver.wave_out_play = alsa_play;
        alsa_driver.name = xstrdup("alsa");
        alsa_driver.description = xstrdup("ALSA output driver, default device: " DEFAULTDEVICE);
        alsa_driver.need_byteswap_on_be = 0;
        alsa_driver.next = NULL;

        if (options)
        {
                pcm_name = xstrdup(options);
        }
        else
        {
                pcm_name = xstrdup(DEFAULTDEVICE);
        }

        return &alsa_driver;
}
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 "rdpsnd.h"
25	#include <unistd.h>
26	#include <fcntl.h>
27	#include <errno.h>
28	#include <alsa/asoundlib.h>
29	#include <sys/time.h>
30
31	#define DEFAULTDEVICE "default"
32	#define MAX_FRAMES 32
33
34	static snd_pcm_t *pcm_handle = NULL;
35	static snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;
36	static BOOL reopened;
37	static short samplewidth;
38	static int audiochannels;
39	static char *pcm_name;
40
41	BOOL
42	alsa_open(void)
43	{
44	int err;
45
46	if ((err = snd_pcm_open(&pcm_handle, pcm_name, stream, 0)) < 0)
47	{
48	error("snd_pcm_open: %s\n", snd_strerror(err));
49	return False;
50	}
51
52	g_dsp_fd = 0;
53	rdpsnd_queue_init();
54
55	reopened = True;
56
57	return True;
58	}
59
60	void
61	alsa_close(void)
62	{
63	/* Ack all remaining packets */
64	while (!rdpsnd_queue_empty())
65	{
66	rdpsnd_send_completion(rdpsnd_queue_current_packet()->tick,
67	rdpsnd_queue_current_packet()->index);
68	rdpsnd_queue_next();
69	}
70
71	if (pcm_handle)
72	{
73	snd_pcm_drop(pcm_handle);
74	snd_pcm_close(pcm_handle);
75	}
76	}
77
78	BOOL
79	alsa_format_supported(WAVEFORMATEX * pwfx)
80	{
81	#if 0
82	int err;
83	snd_pcm_hw_params_t *hwparams = NULL;
84
85	if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
86	{
87	error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
88	return False;
89	}
90
91	if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
92	{
93	error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
94	return False;
95	}
96	snd_pcm_hw_params_free(hwparams);
97	#endif
98
99	if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
100	return False;
101	if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
102	return False;
103	if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
104	return False;
105	if ((pwfx->nSamplesPerSec != 44100) && (pwfx->nSamplesPerSec != 22050))
106	return False;
107
108	return True;
109	}
110
111	BOOL
112	alsa_set_format(WAVEFORMATEX * pwfx)
113	{
114	snd_pcm_hw_params_t *hwparams = NULL;
115	unsigned int rate, exact_rate;
116	int err;
117	unsigned int buffertime;
118
119	samplewidth = pwfx->wBitsPerSample / 8;
120
121	if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
122	{
123	error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
124	return False;
125	}
126
127	if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
128	{
129	error("snd_pcm_hw_params_any: %s\n", snd_strerror(err));
130	return False;
131	}
132
133	if ((err =
134	snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
135	{
136	error("snd_pcm_hw_params_set_access: %s\n", snd_strerror(err));
137	return False;
138	}
139
140	if (pwfx->wBitsPerSample == 16)
141	{
142	if ((err =
143	snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
144	{
145	error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
146	return False;
147	}
148	}
149	else
150	{
151	if ((err =
152	snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S8)) < 0)
153	{
154	error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
155	return False;
156	}
157	}
158
159	#if 0
160	if ((err = snd_pcm_hw_params_set_rate_resample(pcm_handle, hwparams, 1)) < 0)
161	{
162	error("snd_pcm_hw_params_set_rate_resample: %s\n", snd_strerror(err));
163	return False;
164	}
165	#endif
166
167	exact_rate = rate = pwfx->nSamplesPerSec;
168	if ((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_rate, 0)) < 0)
169	{
170	error("snd_pcm_hw_params_set_rate_near: %s\n", snd_strerror(err));
171	return False;
172	}
173
174	audiochannels = pwfx->nChannels;
175	if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, pwfx->nChannels)) < 0)
176	{
177	error("snd_pcm_hw_params_set_channels: %s\n", snd_strerror(err));
178	return False;
179	}
180
181
182	buffertime = 500000; /* microseconds */
183	if ((err =
184	snd_pcm_hw_params_set_buffer_time_near(pcm_handle, hwparams, &buffertime, 0)) < 0)
185	{
186	error("snd_pcm_hw_params_set_buffer_time_near: %s\n", snd_strerror(err));
187	return False;
188	}
189
190	if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0)
191	{
192	error("snd_pcm_hw_params: %s\n", snd_strerror(err));
193	return False;
194	}
195
196	snd_pcm_hw_params_free(hwparams);
197
198	if ((err = snd_pcm_prepare(pcm_handle)) < 0)
199	{
200	error("snd_pcm_prepare: %s\n", snd_strerror(err));
201	return False;
202	}
203
204	reopened = True;
205
206	return True;
207	}
208
209	void
210	alsa_volume(uint16 left, uint16 right)
211	{
212	static int warned = 0;
213
214	if (!warned)
215	{
216	warning("volume changes currently not supported with alsa-output\n");
217	warned = 1;
218	}
219	}
220
221	void
222	alsa_play(void)
223	{
224	struct audio_packet *packet;
225	STREAM out;
226	int len;
227	static long prev_s, prev_us;
228	unsigned int duration;
229	struct timeval tv;
230	int next_tick;
231
232	if (reopened)
233	{
234	reopened = False;
235	gettimeofday(&tv, NULL);
236	prev_s = tv.tv_sec;
237	prev_us = tv.tv_usec;
238	}
239
240	if (rdpsnd_queue_empty())
241	{
242	g_dsp_busy = 0;
243	return;
244	}
245
246	packet = rdpsnd_queue_current_packet();
247	out = &packet->s;
248
249	next_tick = rdpsnd_queue_next_tick();
250
251	len = (out->end - out->p) / (samplewidth * audiochannels);
252	if ((len = snd_pcm_writei(pcm_handle, out->p, ((MAX_FRAMES < len) ? MAX_FRAMES : len))) < 0)
253	{
254	snd_pcm_prepare(pcm_handle);
255	len = 0;
256	}
257	out->p += (len * samplewidth * audiochannels);
258
259	gettimeofday(&tv, NULL);
260
261	duration = ((tv.tv_sec - prev_s) * 1000000 + (tv.tv_usec - prev_us)) / 1000;
262
263	if (packet->tick > next_tick)
264	next_tick += 65536;
265
266	if ((out->p == out->end) \|\| duration > next_tick - packet->tick + 500)
267	{
268	prev_s = tv.tv_sec;
269	prev_us = tv.tv_usec;
270
271	if (abs((next_tick - packet->tick) - duration) > 20)
272	{
273	DEBUG(("duration: %d, calc: %d, ", duration, next_tick - packet->tick));
274	DEBUG(("last: %d, is: %d, should: %d\n", packet->tick,
275	(packet->tick + duration) % 65536, next_tick % 65536));
276	}
277
278	rdpsnd_send_completion(((packet->tick + duration) % 65536), packet->index);
279	rdpsnd_queue_next();
280	}
281
282	g_dsp_busy = 1;
283	return;
284	}
285
286	struct audio_driver *
287	alsa_register(char *options)
288	{
289	static struct audio_driver alsa_driver;
290
291	alsa_driver.wave_out_write = rdpsnd_queue_write;
292	alsa_driver.wave_out_open = alsa_open;
293	alsa_driver.wave_out_close = alsa_close;
294	alsa_driver.wave_out_format_supported = alsa_format_supported;
295	alsa_driver.wave_out_set_format = alsa_set_format;
296	alsa_driver.wave_out_volume = alsa_volume;
297	alsa_driver.wave_out_play = alsa_play;
298	alsa_driver.name = xstrdup("alsa");
299	alsa_driver.description = xstrdup("ALSA output driver, default device: " DEFAULTDEVICE);
300	alsa_driver.need_byteswap_on_be = 0;
301	alsa_driver.next = NULL;
302
303	if (options)
304	{
305	pcm_name = xstrdup(options);
306	}
307	else
308	{
309	pcm_name = xstrdup(DEFAULTDEVICE);
310	}
311
312	return &alsa_driver;
313	}