rdesktop/trunk/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);
        }
}

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	stargo	1253	/* -- 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	stargo	1254	#include "rdpsnd.h"
25	stargo	1261	#include "rdpsnd_dsp.h"
26	stargo	1253	#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	ossman_	1302	static unsigned int rate;
41	stargo	1255	static char *pcm_name;
42	stargo	1253
43			BOOL
44	stargo	1255	alsa_open(void)
45	stargo	1253	{
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	stargo	1255	alsa_close(void)
63	stargo	1253	{
64			/* Ack all remaining packets */
65	stargo	1254	while (!rdpsnd_queue_empty())
66	ossman_	1302	rdpsnd_queue_next(0);
67	stargo	1253
68			if (pcm_handle)
69			{
70			snd_pcm_drop(pcm_handle);
71			snd_pcm_close(pcm_handle);
72			}
73			}
74
75			BOOL
76	stargo	1255	alsa_format_supported(WAVEFORMATEX * pwfx)
77	stargo	1253	{
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	stargo	1255	alsa_set_format(WAVEFORMATEX * pwfx)
110	stargo	1253	{
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	ossman_	1302	rate = pwfx->nSamplesPerSec;
164			if ((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &rate, 0)) < 0)
165	stargo	1253	{
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	stargo	1255	alsa_play(void)
207	stargo	1253	{
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	stargo	1254	if (rdpsnd_queue_empty())
225	stargo	1253	{
226			g_dsp_busy = 0;
227			return;
228			}
229
230	stargo	1254	packet = rdpsnd_queue_current_packet();
231	stargo	1253	out = &packet->s;
232
233	stargo	1254	next_tick = rdpsnd_queue_next_tick();
234	stargo	1253
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	ossman_	1302	snd_pcm_sframes_t delay_frames;
253			unsigned long delay_us;
254
255	stargo	1253	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	ossman_	1302	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	stargo	1253	}
274
275			g_dsp_busy = 1;
276			return;
277			}
278	stargo	1255
279			struct audio_driver *
280			alsa_register(char *options)
281			{
282	stargo	1256	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	stargo	1261	alsa_driver.wave_out_volume = rdpsnd_dsp_softvol_set;
289	stargo	1256	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	stargo	1260	alsa_driver.need_byteswap_on_be = 0;
293	stargo	1279	alsa_driver.need_resampling = 0;
294	stargo	1256	alsa_driver.next = NULL;
295
296	stargo	1255	if (options)
297			{
298			pcm_name = xstrdup(options);
299			}
300			else
301			{
302			pcm_name = xstrdup(DEFAULTDEVICE);
303			}
304
305			return &alsa_driver;
306			}