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 unsigned int rate;
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;

        reopened = True;

        return True;
}

void
alsa_close(void)
{
        /* Ack all remaining packets */
        while (!rdpsnd_queue_empty())
                rdpsnd_queue_next(0);

        if (pcm_handle)
        {
                snd_pcm_close(pcm_handle);
                pcm_handle = NULL;
        }
}

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;
        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

        rate = pwfx->nSamplesPerSec;
        if ((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &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)
        {
                snd_pcm_sframes_t delay_frames;
                unsigned long delay_us;

                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));
                }

                if (snd_pcm_delay(pcm_handle, &delay_frames) < 0)
                        delay_frames = out->size / (samplewidth * audiochannels);
                if (delay_frames < 0)
                        delay_frames = 0;

                delay_us = delay_frames * (1000000 / rate);

                rdpsnd_queue_next(delay_us);
        }

        g_dsp_busy = 1;
        return;
}

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

        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 unsigned int rate;
41	static char *pcm_name;
42
43	BOOL
44	alsa_open(void)
45	{
46	int err;
47
48	if ((err = snd_pcm_open(&pcm_handle, pcm_name, stream, 0)) < 0)
49	{
50	error("snd_pcm_open: %s\n", snd_strerror(err));
51	return False;
52	}
53
54	g_dsp_fd = 0;
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	rdpsnd_queue_next(0);
67
68	if (pcm_handle)
69	{
70	snd_pcm_close(pcm_handle);
71	pcm_handle = NULL;
72	}
73	}
74
75	BOOL
76	alsa_format_supported(WAVEFORMATEX * pwfx)
77	{
78	#if 0
79	int err;
80	snd_pcm_hw_params_t *hwparams = NULL;
81
82	if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
83	{
84	error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
85	return False;
86	}
87
88	if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
89	{
90	error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
91	return False;
92	}
93	snd_pcm_hw_params_free(hwparams);
94	#endif
95
96	if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
97	return False;
98	if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
99	return False;
100	if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
101	return False;
102	if ((pwfx->nSamplesPerSec != 44100) && (pwfx->nSamplesPerSec != 22050))
103	return False;
104
105	return True;
106	}
107
108	BOOL
109	alsa_set_format(WAVEFORMATEX * pwfx)
110	{
111	snd_pcm_hw_params_t *hwparams = NULL;
112	int err;
113	unsigned int buffertime;
114
115	samplewidth = pwfx->wBitsPerSample / 8;
116
117	if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
118	{
119	error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
120	return False;
121	}
122
123	if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
124	{
125	error("snd_pcm_hw_params_any: %s\n", snd_strerror(err));
126	return False;
127	}
128
129	if ((err =
130	snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
131	{
132	error("snd_pcm_hw_params_set_access: %s\n", snd_strerror(err));
133	return False;
134	}
135
136	if (pwfx->wBitsPerSample == 16)
137	{
138	if ((err =
139	snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
140	{
141	error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
142	return False;
143	}
144	}
145	else
146	{
147	if ((err =
148	snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S8)) < 0)
149	{
150	error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
151	return False;
152	}
153	}
154
155	#if 0
156	if ((err = snd_pcm_hw_params_set_rate_resample(pcm_handle, hwparams, 1)) < 0)
157	{
158	error("snd_pcm_hw_params_set_rate_resample: %s\n", snd_strerror(err));
159	return False;
160	}
161	#endif
162
163	rate = pwfx->nSamplesPerSec;
164	if ((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &rate, 0)) < 0)
165	{
166	error("snd_pcm_hw_params_set_rate_near: %s\n", snd_strerror(err));
167	return False;
168	}
169
170	audiochannels = pwfx->nChannels;
171	if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, pwfx->nChannels)) < 0)
172	{
173	error("snd_pcm_hw_params_set_channels: %s\n", snd_strerror(err));
174	return False;
175	}
176
177
178	buffertime = 500000; /* microseconds */
179	if ((err =
180	snd_pcm_hw_params_set_buffer_time_near(pcm_handle, hwparams, &buffertime, 0)) < 0)
181	{
182	error("snd_pcm_hw_params_set_buffer_time_near: %s\n", snd_strerror(err));
183	return False;
184	}
185
186	if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0)
187	{
188	error("snd_pcm_hw_params: %s\n", snd_strerror(err));
189	return False;
190	}
191
192	snd_pcm_hw_params_free(hwparams);
193
194	if ((err = snd_pcm_prepare(pcm_handle)) < 0)
195	{
196	error("snd_pcm_prepare: %s\n", snd_strerror(err));
197	return False;
198	}
199
200	reopened = True;
201
202	return True;
203	}
204
205	void
206	alsa_play(void)
207	{
208	struct audio_packet *packet;
209	STREAM out;
210	int len;
211	static long prev_s, prev_us;
212	unsigned int duration;
213	struct timeval tv;
214	int next_tick;
215
216	if (reopened)
217	{
218	reopened = False;
219	gettimeofday(&tv, NULL);
220	prev_s = tv.tv_sec;
221	prev_us = tv.tv_usec;
222	}
223
224	if (rdpsnd_queue_empty())
225	{
226	g_dsp_busy = 0;
227	return;
228	}
229
230	packet = rdpsnd_queue_current_packet();
231	out = &packet->s;
232
233	next_tick = rdpsnd_queue_next_tick();
234
235	len = (out->end - out->p) / (samplewidth * audiochannels);
236	if ((len = snd_pcm_writei(pcm_handle, out->p, ((MAX_FRAMES < len) ? MAX_FRAMES : len))) < 0)
237	{
238	snd_pcm_prepare(pcm_handle);
239	len = 0;
240	}
241	out->p += (len * samplewidth * audiochannels);
242
243	gettimeofday(&tv, NULL);
244
245	duration = ((tv.tv_sec - prev_s) * 1000000 + (tv.tv_usec - prev_us)) / 1000;
246
247	if (packet->tick > next_tick)
248	next_tick += 65536;
249
250	if ((out->p == out->end) \|\| duration > next_tick - packet->tick + 500)
251	{
252	snd_pcm_sframes_t delay_frames;
253	unsigned long delay_us;
254
255	prev_s = tv.tv_sec;
256	prev_us = tv.tv_usec;
257
258	if (abs((next_tick - packet->tick) - duration) > 20)
259	{
260	DEBUG(("duration: %d, calc: %d, ", duration, next_tick - packet->tick));
261	DEBUG(("last: %d, is: %d, should: %d\n", packet->tick,
262	(packet->tick + duration) % 65536, next_tick % 65536));
263	}
264
265	if (snd_pcm_delay(pcm_handle, &delay_frames) < 0)
266	delay_frames = out->size / (samplewidth * audiochannels);
267	if (delay_frames < 0)
268	delay_frames = 0;
269
270	delay_us = delay_frames * (1000000 / rate);
271
272	rdpsnd_queue_next(delay_us);
273	}
274
275	g_dsp_busy = 1;
276	return;
277	}
278
279	struct audio_driver *
280	alsa_register(char *options)
281	{
282	static struct audio_driver alsa_driver;
283
284	alsa_driver.wave_out_open = alsa_open;
285	alsa_driver.wave_out_close = alsa_close;
286	alsa_driver.wave_out_format_supported = alsa_format_supported;
287	alsa_driver.wave_out_set_format = alsa_set_format;
288	alsa_driver.wave_out_volume = rdpsnd_dsp_softvol_set;
289	alsa_driver.wave_out_play = alsa_play;
290	alsa_driver.name = xstrdup("alsa");
291	alsa_driver.description = xstrdup("ALSA output driver, default device: " DEFAULTDEVICE);
292	alsa_driver.need_byteswap_on_be = 0;
293	alsa_driver.need_resampling = 0;
294	alsa_driver.next = NULL;
295
296	if (options)
297	{
298	pcm_name = xstrdup(options);
299	}
300	else
301	{
302	pcm_name = xstrdup(DEFAULTDEVICE);
303	}
304
305	return &alsa_driver;
306	}