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	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	stargo	1325	pcm_handle = NULL;
73	stargo	1253	}
74			}
75
76			BOOL
77	stargo	1255	alsa_format_supported(WAVEFORMATEX * pwfx)
78	stargo	1253	{
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	stargo	1255	alsa_set_format(WAVEFORMATEX * pwfx)
111	stargo	1253	{
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	ossman_	1302	rate = pwfx->nSamplesPerSec;
165			if ((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &rate, 0)) < 0)
166	stargo	1253	{
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	stargo	1255	alsa_play(void)
208	stargo	1253	{
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	stargo	1254	if (rdpsnd_queue_empty())
226	stargo	1253	{
227			g_dsp_busy = 0;
228			return;
229			}
230
231	stargo	1254	packet = rdpsnd_queue_current_packet();
232	stargo	1253	out = &packet->s;
233
234	stargo	1254	next_tick = rdpsnd_queue_next_tick();
235	stargo	1253
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	ossman_	1302	snd_pcm_sframes_t delay_frames;
254			unsigned long delay_us;
255
256	stargo	1253	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	ossman_	1302	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	stargo	1253	}
275
276			g_dsp_busy = 1;
277			return;
278			}
279	stargo	1255
280			struct audio_driver *
281			alsa_register(char *options)
282			{
283	stargo	1256	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	stargo	1261	alsa_driver.wave_out_volume = rdpsnd_dsp_softvol_set;
290	stargo	1256	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	stargo	1260	alsa_driver.need_byteswap_on_be = 0;
294	stargo	1279	alsa_driver.need_resampling = 0;
295	stargo	1256	alsa_driver.next = NULL;
296
297	stargo	1255	if (options)
298			{
299			pcm_name = xstrdup(options);
300			}
301			else
302			{
303			pcm_name = xstrdup(DEFAULTDEVICE);
304			}
305
306			return &alsa_driver;
307			}