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 "rdpsnd_dsp.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_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 = rdpsnd_dsp_softvol_set;
        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.need_resampling = 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 "rdpsnd_dsp.h"
26	#include <unistd.h>
27	#include <fcntl.h>
28	#include <errno.h>
29	#include <alsa/asoundlib.h>
30	#include <sys/time.h>
31
32	#define DEFAULTDEVICE "default"
33	#define MAX_FRAMES 32
34
35	static snd_pcm_t *pcm_handle = NULL;
36	static snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;
37	static BOOL reopened;
38	static short samplewidth;
39	static int audiochannels;
40	static char *pcm_name;
41
42	BOOL
43	alsa_open(void)
44	{
45	int err;
46
47	if ((err = snd_pcm_open(&pcm_handle, pcm_name, stream, 0)) < 0)
48	{
49	error("snd_pcm_open: %s\n", snd_strerror(err));
50	return False;
51	}
52
53	g_dsp_fd = 0;
54	rdpsnd_queue_init();
55
56	reopened = True;
57
58	return True;
59	}
60
61	void
62	alsa_close(void)
63	{
64	/* Ack all remaining packets */
65	while (!rdpsnd_queue_empty())
66	{
67	rdpsnd_send_completion(rdpsnd_queue_current_packet()->tick,
68	rdpsnd_queue_current_packet()->index);
69	rdpsnd_queue_next();
70	}
71
72	if (pcm_handle)
73	{
74	snd_pcm_drop(pcm_handle);
75	snd_pcm_close(pcm_handle);
76	}
77	}
78
79	BOOL
80	alsa_format_supported(WAVEFORMATEX * pwfx)
81	{
82	#if 0
83	int err;
84	snd_pcm_hw_params_t *hwparams = NULL;
85
86	if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
87	{
88	error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
89	return False;
90	}
91
92	if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
93	{
94	error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
95	return False;
96	}
97	snd_pcm_hw_params_free(hwparams);
98	#endif
99
100	if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
101	return False;
102	if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
103	return False;
104	if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
105	return False;
106	if ((pwfx->nSamplesPerSec != 44100) && (pwfx->nSamplesPerSec != 22050))
107	return False;
108
109	return True;
110	}
111
112	BOOL
113	alsa_set_format(WAVEFORMATEX * pwfx)
114	{
115	snd_pcm_hw_params_t *hwparams = NULL;
116	unsigned int rate, exact_rate;
117	int err;
118	unsigned int buffertime;
119
120	samplewidth = pwfx->wBitsPerSample / 8;
121
122	if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
123	{
124	error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
125	return False;
126	}
127
128	if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
129	{
130	error("snd_pcm_hw_params_any: %s\n", snd_strerror(err));
131	return False;
132	}
133
134	if ((err =
135	snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
136	{
137	error("snd_pcm_hw_params_set_access: %s\n", snd_strerror(err));
138	return False;
139	}
140
141	if (pwfx->wBitsPerSample == 16)
142	{
143	if ((err =
144	snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
145	{
146	error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
147	return False;
148	}
149	}
150	else
151	{
152	if ((err =
153	snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S8)) < 0)
154	{
155	error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
156	return False;
157	}
158	}
159
160	#if 0
161	if ((err = snd_pcm_hw_params_set_rate_resample(pcm_handle, hwparams, 1)) < 0)
162	{
163	error("snd_pcm_hw_params_set_rate_resample: %s\n", snd_strerror(err));
164	return False;
165	}
166	#endif
167
168	exact_rate = rate = pwfx->nSamplesPerSec;
169	if ((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_rate, 0)) < 0)
170	{
171	error("snd_pcm_hw_params_set_rate_near: %s\n", snd_strerror(err));
172	return False;
173	}
174
175	audiochannels = pwfx->nChannels;
176	if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, pwfx->nChannels)) < 0)
177	{
178	error("snd_pcm_hw_params_set_channels: %s\n", snd_strerror(err));
179	return False;
180	}
181
182
183	buffertime = 500000; /* microseconds */
184	if ((err =
185	snd_pcm_hw_params_set_buffer_time_near(pcm_handle, hwparams, &buffertime, 0)) < 0)
186	{
187	error("snd_pcm_hw_params_set_buffer_time_near: %s\n", snd_strerror(err));
188	return False;
189	}
190
191	if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0)
192	{
193	error("snd_pcm_hw_params: %s\n", snd_strerror(err));
194	return False;
195	}
196
197	snd_pcm_hw_params_free(hwparams);
198
199	if ((err = snd_pcm_prepare(pcm_handle)) < 0)
200	{
201	error("snd_pcm_prepare: %s\n", snd_strerror(err));
202	return False;
203	}
204
205	reopened = True;
206
207	return True;
208	}
209
210	void
211	alsa_play(void)
212	{
213	struct audio_packet *packet;
214	STREAM out;
215	int len;
216	static long prev_s, prev_us;
217	unsigned int duration;
218	struct timeval tv;
219	int next_tick;
220
221	if (reopened)
222	{
223	reopened = False;
224	gettimeofday(&tv, NULL);
225	prev_s = tv.tv_sec;
226	prev_us = tv.tv_usec;
227	}
228
229	if (rdpsnd_queue_empty())
230	{
231	g_dsp_busy = 0;
232	return;
233	}
234
235	packet = rdpsnd_queue_current_packet();
236	out = &packet->s;
237
238	next_tick = rdpsnd_queue_next_tick();
239
240	len = (out->end - out->p) / (samplewidth * audiochannels);
241	if ((len = snd_pcm_writei(pcm_handle, out->p, ((MAX_FRAMES < len) ? MAX_FRAMES : len))) < 0)
242	{
243	snd_pcm_prepare(pcm_handle);
244	len = 0;
245	}
246	out->p += (len * samplewidth * audiochannels);
247
248	gettimeofday(&tv, NULL);
249
250	duration = ((tv.tv_sec - prev_s) * 1000000 + (tv.tv_usec - prev_us)) / 1000;
251
252	if (packet->tick > next_tick)
253	next_tick += 65536;
254
255	if ((out->p == out->end) \|\| duration > next_tick - packet->tick + 500)
256	{
257	prev_s = tv.tv_sec;
258	prev_us = tv.tv_usec;
259
260	if (abs((next_tick - packet->tick) - duration) > 20)
261	{
262	DEBUG(("duration: %d, calc: %d, ", duration, next_tick - packet->tick));
263	DEBUG(("last: %d, is: %d, should: %d\n", packet->tick,
264	(packet->tick + duration) % 65536, next_tick % 65536));
265	}
266
267	rdpsnd_send_completion(((packet->tick + duration) % 65536), packet->index);
268	rdpsnd_queue_next();
269	}
270
271	g_dsp_busy = 1;
272	return;
273	}
274
275	struct audio_driver *
276	alsa_register(char *options)
277	{
278	static struct audio_driver alsa_driver;
279
280	alsa_driver.wave_out_write = rdpsnd_queue_write;
281	alsa_driver.wave_out_open = alsa_open;
282	alsa_driver.wave_out_close = alsa_close;
283	alsa_driver.wave_out_format_supported = alsa_format_supported;
284	alsa_driver.wave_out_set_format = alsa_set_format;
285	alsa_driver.wave_out_volume = rdpsnd_dsp_softvol_set;
286	alsa_driver.wave_out_play = alsa_play;
287	alsa_driver.name = xstrdup("alsa");
288	alsa_driver.description = xstrdup("ALSA output driver, default device: " DEFAULTDEVICE);
289	alsa_driver.need_byteswap_on_be = 0;
290	alsa_driver.need_resampling = 0;
291	alsa_driver.next = NULL;
292
293	if (options)
294	{
295	pcm_name = xstrdup(options);
296	}
297	else
298	{
299	pcm_name = xstrdup(DEFAULTDEVICE);
300	}
301
302	return &alsa_driver;
303	}