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_drop(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_drop(pcm_handle);
71	snd_pcm_close(pcm_handle);
72	pcm_handle = NULL;
73	}
74	}
75
76	BOOL
77	alsa_format_supported(WAVEFORMATEX * pwfx)
78	{
79	#if 0
80	int err;
81	snd_pcm_hw_params_t *hwparams = NULL;
82
83	if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
84	{
85	error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
86	return False;
87	}
88
89	if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
90	{
91	error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
92	return False;
93	}
94	snd_pcm_hw_params_free(hwparams);
95	#endif
96
97	if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
98	return False;
99	if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
100	return False;
101	if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
102	return False;
103	if ((pwfx->nSamplesPerSec != 44100) && (pwfx->nSamplesPerSec != 22050))
104	return False;
105
106	return True;
107	}
108
109	BOOL
110	alsa_set_format(WAVEFORMATEX * pwfx)
111	{
112	snd_pcm_hw_params_t *hwparams = NULL;
113	int err;
114	unsigned int buffertime;
115
116	samplewidth = pwfx->wBitsPerSample / 8;
117
118	if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
119	{
120	error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
121	return False;
122	}
123
124	if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
125	{
126	error("snd_pcm_hw_params_any: %s\n", snd_strerror(err));
127	return False;
128	}
129
130	if ((err =
131	snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
132	{
133	error("snd_pcm_hw_params_set_access: %s\n", snd_strerror(err));
134	return False;
135	}
136
137	if (pwfx->wBitsPerSample == 16)
138	{
139	if ((err =
140	snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
141	{
142	error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
143	return False;
144	}
145	}
146	else
147	{
148	if ((err =
149	snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S8)) < 0)
150	{
151	error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
152	return False;
153	}
154	}
155
156	#if 0
157	if ((err = snd_pcm_hw_params_set_rate_resample(pcm_handle, hwparams, 1)) < 0)
158	{
159	error("snd_pcm_hw_params_set_rate_resample: %s\n", snd_strerror(err));
160	return False;
161	}
162	#endif
163
164	rate = pwfx->nSamplesPerSec;
165	if ((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &rate, 0)) < 0)
166	{
167	error("snd_pcm_hw_params_set_rate_near: %s\n", snd_strerror(err));
168	return False;
169	}
170
171	audiochannels = pwfx->nChannels;
172	if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, pwfx->nChannels)) < 0)
173	{
174	error("snd_pcm_hw_params_set_channels: %s\n", snd_strerror(err));
175	return False;
176	}
177
178
179	buffertime = 500000; /* microseconds */
180	if ((err =
181	snd_pcm_hw_params_set_buffer_time_near(pcm_handle, hwparams, &buffertime, 0)) < 0)
182	{
183	error("snd_pcm_hw_params_set_buffer_time_near: %s\n", snd_strerror(err));
184	return False;
185	}
186
187	if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0)
188	{
189	error("snd_pcm_hw_params: %s\n", snd_strerror(err));
190	return False;
191	}
192
193	snd_pcm_hw_params_free(hwparams);
194
195	if ((err = snd_pcm_prepare(pcm_handle)) < 0)
196	{
197	error("snd_pcm_prepare: %s\n", snd_strerror(err));
198	return False;
199	}
200
201	reopened = True;
202
203	return True;
204	}
205
206	void
207	alsa_play(void)
208	{
209	struct audio_packet *packet;
210	STREAM out;
211	int len;
212	static long prev_s, prev_us;
213	unsigned int duration;
214	struct timeval tv;
215	int next_tick;
216
217	if (reopened)
218	{
219	reopened = False;
220	gettimeofday(&tv, NULL);
221	prev_s = tv.tv_sec;
222	prev_us = tv.tv_usec;
223	}
224
225	if (rdpsnd_queue_empty())
226	{
227	g_dsp_busy = 0;
228	return;
229	}
230
231	packet = rdpsnd_queue_current_packet();
232	out = &packet->s;
233
234	next_tick = rdpsnd_queue_next_tick();
235
236	len = (out->end - out->p) / (samplewidth * audiochannels);
237	if ((len = snd_pcm_writei(pcm_handle, out->p, ((MAX_FRAMES < len) ? MAX_FRAMES : len))) < 0)
238	{
239	snd_pcm_prepare(pcm_handle);
240	len = 0;
241	}
242	out->p += (len * samplewidth * audiochannels);
243
244	gettimeofday(&tv, NULL);
245
246	duration = ((tv.tv_sec - prev_s) * 1000000 + (tv.tv_usec - prev_us)) / 1000;
247
248	if (packet->tick > next_tick)
249	next_tick += 65536;
250
251	if ((out->p == out->end) \|\| duration > next_tick - packet->tick + 500)
252	{
253	snd_pcm_sframes_t delay_frames;
254	unsigned long delay_us;
255
256	prev_s = tv.tv_sec;
257	prev_us = tv.tv_usec;
258
259	if (abs((next_tick - packet->tick) - duration) > 20)
260	{
261	DEBUG(("duration: %d, calc: %d, ", duration, next_tick - packet->tick));
262	DEBUG(("last: %d, is: %d, should: %d\n", packet->tick,
263	(packet->tick + duration) % 65536, next_tick % 65536));
264	}
265
266	if (snd_pcm_delay(pcm_handle, &delay_frames) < 0)
267	delay_frames = out->size / (samplewidth * audiochannels);
268	if (delay_frames < 0)
269	delay_frames = 0;
270
271	delay_us = delay_frames * (1000000 / rate);
272
273	rdpsnd_queue_next(delay_us);
274	}
275
276	g_dsp_busy = 1;
277	return;
278	}
279
280	struct audio_driver *
281	alsa_register(char *options)
282	{
283	static struct audio_driver alsa_driver;
284
285	alsa_driver.wave_out_open = alsa_open;
286	alsa_driver.wave_out_close = alsa_close;
287	alsa_driver.wave_out_format_supported = alsa_format_supported;
288	alsa_driver.wave_out_set_format = alsa_set_format;
289	alsa_driver.wave_out_volume = rdpsnd_dsp_softvol_set;
290	alsa_driver.wave_out_play = alsa_play;
291	alsa_driver.name = xstrdup("alsa");
292	alsa_driver.description = xstrdup("ALSA output driver, default device: " DEFAULTDEVICE);
293	alsa_driver.need_byteswap_on_be = 0;
294	alsa_driver.need_resampling = 0;
295	alsa_driver.next = NULL;
296
297	if (options)
298	{
299	pcm_name = xstrdup(options);
300	}
301	else
302	{
303	pcm_name = xstrdup(DEFAULTDEVICE);
304	}
305
306	return &alsa_driver;
307	}