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;
}

static struct audio_driver alsa_driver = {
      wave_out_write:rdpsnd_queue_write,
      wave_out_open:alsa_open,
      wave_out_close:alsa_close,
      wave_out_format_supported:alsa_format_supported,
      wave_out_set_format:alsa_set_format,
      wave_out_volume:alsa_volume,
      wave_out_play:alsa_play,
      name:"alsa",
      description:"ALSA output driver, default device: " DEFAULTDEVICE,
      next:NULL,
};

struct audio_driver *
alsa_register(char *options)
{
        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	1253	#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	stargo	1255	static char *pcm_name;
40	stargo	1253
41			BOOL
42	stargo	1255	alsa_open(void)
43	stargo	1253	{
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	stargo	1254	rdpsnd_queue_init();
54	stargo	1253
55			reopened = True;
56
57			return True;
58			}
59
60			void
61	stargo	1255	alsa_close(void)
62	stargo	1253	{
63			/* Ack all remaining packets */
64	stargo	1254	while (!rdpsnd_queue_empty())
65	stargo	1253	{
66	stargo	1254	rdpsnd_send_completion(rdpsnd_queue_current_packet()->tick,
67			rdpsnd_queue_current_packet()->index);
68			rdpsnd_queue_next();
69	stargo	1253	}
70
71			if (pcm_handle)
72			{
73			snd_pcm_drop(pcm_handle);
74			snd_pcm_close(pcm_handle);
75			}
76			}
77
78			BOOL
79	stargo	1255	alsa_format_supported(WAVEFORMATEX * pwfx)
80	stargo	1253	{
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	stargo	1255	alsa_set_format(WAVEFORMATEX * pwfx)
113	stargo	1253	{
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	stargo	1255	alsa_volume(uint16 left, uint16 right)
211	stargo	1253	{
212			static int warned = 0;
213
214			if (!warned)
215			{
216	stargo	1255	warning("volume changes currently not supported with alsa-output\n");
217	stargo	1253	warned = 1;
218			}
219			}
220
221			void
222	stargo	1255	alsa_play(void)
223	stargo	1253	{
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	stargo	1254	if (rdpsnd_queue_empty())
241	stargo	1253	{
242			g_dsp_busy = 0;
243			return;
244			}
245
246	stargo	1254	packet = rdpsnd_queue_current_packet();
247	stargo	1253	out = &packet->s;
248
249	stargo	1254	next_tick = rdpsnd_queue_next_tick();
250	stargo	1253
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	stargo	1254	rdpsnd_send_completion(((packet->tick + duration) % 65536), packet->index);
279			rdpsnd_queue_next();
280	stargo	1253	}
281
282			g_dsp_busy = 1;
283			return;
284			}
285	stargo	1255
286			static struct audio_driver alsa_driver = {
287			wave_out_write:rdpsnd_queue_write,
288			wave_out_open:alsa_open,
289			wave_out_close:alsa_close,
290			wave_out_format_supported:alsa_format_supported,
291			wave_out_set_format:alsa_set_format,
292			wave_out_volume:alsa_volume,
293			wave_out_play:alsa_play,
294			name:"alsa",
295			description:"ALSA output driver, default device: " DEFAULTDEVICE,
296			next:NULL,
297			};
298
299			struct audio_driver *
300			alsa_register(char *options)
301			{
302			if (options)
303			{
304			pcm_name = xstrdup(options);
305			}
306			else
307			{
308			pcm_name = xstrdup(DEFAULTDEVICE);
309			}
310
311			return &alsa_driver;
312			}