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	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	stargo	1255	static char *pcm_name;
41	stargo	1253
42			BOOL
43	stargo	1255	alsa_open(void)
44	stargo	1253	{
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	stargo	1254	rdpsnd_queue_init();
55	stargo	1253
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	stargo	1253	{
67	stargo	1254	rdpsnd_send_completion(rdpsnd_queue_current_packet()->tick,
68			rdpsnd_queue_current_packet()->index);
69			rdpsnd_queue_next();
70	stargo	1253	}
71
72			if (pcm_handle)
73			{
74			snd_pcm_drop(pcm_handle);
75			snd_pcm_close(pcm_handle);
76			}
77			}
78
79			BOOL
80	stargo	1255	alsa_format_supported(WAVEFORMATEX * pwfx)
81	stargo	1253	{
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	stargo	1255	alsa_set_format(WAVEFORMATEX * pwfx)
114	stargo	1253	{
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	stargo	1255	alsa_play(void)
212	stargo	1253	{
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	stargo	1254	if (rdpsnd_queue_empty())
230	stargo	1253	{
231			g_dsp_busy = 0;
232			return;
233			}
234
235	stargo	1254	packet = rdpsnd_queue_current_packet();
236	stargo	1253	out = &packet->s;
237
238	stargo	1254	next_tick = rdpsnd_queue_next_tick();
239	stargo	1253
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	stargo	1254	rdpsnd_send_completion(((packet->tick + duration) % 65536), packet->index);
268			rdpsnd_queue_next();
269	stargo	1253	}
270
271			g_dsp_busy = 1;
272			return;
273			}
274	stargo	1255
275			struct audio_driver *
276			alsa_register(char *options)
277			{
278	stargo	1256	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	stargo	1261	alsa_driver.wave_out_volume = rdpsnd_dsp_softvol_set;
286	stargo	1256	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	stargo	1260	alsa_driver.need_byteswap_on_be = 0;
290	stargo	1279	alsa_driver.need_resampling = 0;
291	stargo	1256	alsa_driver.next = NULL;
292
293	stargo	1255	if (options)
294			{
295			pcm_name = xstrdup(options);
296			}
297			else
298			{
299			pcm_name = xstrdup(DEFAULTDEVICE);
300			}
301
302			return &alsa_driver;
303			}