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;

BOOL
wave_out_open(void)
{
        char *pcm_name;
        int err;

        pcm_name = xstrdup(DEFAULTDEVICE);

        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
wave_out_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
wave_out_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
wave_out_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
wave_out_volume(uint16 left, uint16 right)
{
        static int warned = 0;

        if (!warned)
        {
                warning("volume changes currently not supported with experimental alsa-output\n");
                warned = 1;
        }
}

void
wave_out_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;
}
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
40			BOOL
41			wave_out_open(void)
42			{
43			char *pcm_name;
44			int err;
45
46			pcm_name = xstrdup(DEFAULTDEVICE);
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	stargo	1254	rdpsnd_queue_init();
56	stargo	1253
57			reopened = True;
58
59			return True;
60			}
61
62			void
63			wave_out_close(void)
64			{
65			/* Ack all remaining packets */
66	stargo	1254	while (!rdpsnd_queue_empty())
67	stargo	1253	{
68	stargo	1254	rdpsnd_send_completion(rdpsnd_queue_current_packet()->tick,
69			rdpsnd_queue_current_packet()->index);
70			rdpsnd_queue_next();
71	stargo	1253	}
72
73	stargo	1254
74	stargo	1253	if (pcm_handle)
75			{
76			snd_pcm_drop(pcm_handle);
77			snd_pcm_close(pcm_handle);
78			}
79			}
80
81			BOOL
82			wave_out_format_supported(WAVEFORMATEX * pwfx)
83			{
84			#if 0
85			int err;
86			snd_pcm_hw_params_t *hwparams = NULL;
87
88			if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
89			{
90			error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
91			return False;
92			}
93
94			if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
95			{
96			error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
97			return False;
98			}
99			snd_pcm_hw_params_free(hwparams);
100			#endif
101
102			if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
103			return False;
104			if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
105			return False;
106			if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
107			return False;
108			if ((pwfx->nSamplesPerSec != 44100) && (pwfx->nSamplesPerSec != 22050))
109			return False;
110
111			return True;
112			}
113
114			BOOL
115			wave_out_set_format(WAVEFORMATEX * pwfx)
116			{
117			snd_pcm_hw_params_t *hwparams = NULL;
118			unsigned int rate, exact_rate;
119			int err;
120			unsigned int buffertime;
121
122			samplewidth = pwfx->wBitsPerSample / 8;
123
124			if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
125			{
126			error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
127			return False;
128			}
129
130			if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
131			{
132			error("snd_pcm_hw_params_any: %s\n", snd_strerror(err));
133			return False;
134			}
135
136			if ((err =
137			snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
138			{
139			error("snd_pcm_hw_params_set_access: %s\n", snd_strerror(err));
140			return False;
141			}
142
143			if (pwfx->wBitsPerSample == 16)
144			{
145			if ((err =
146			snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
147			{
148			error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
149			return False;
150			}
151			}
152			else
153			{
154			if ((err =
155			snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S8)) < 0)
156			{
157			error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
158			return False;
159			}
160			}
161
162			#if 0
163			if ((err = snd_pcm_hw_params_set_rate_resample(pcm_handle, hwparams, 1)) < 0)
164			{
165			error("snd_pcm_hw_params_set_rate_resample: %s\n", snd_strerror(err));
166			return False;
167			}
168			#endif
169
170			exact_rate = rate = pwfx->nSamplesPerSec;
171			if ((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_rate, 0)) < 0)
172			{
173			error("snd_pcm_hw_params_set_rate_near: %s\n", snd_strerror(err));
174			return False;
175			}
176
177			audiochannels = pwfx->nChannels;
178			if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, pwfx->nChannels)) < 0)
179			{
180			error("snd_pcm_hw_params_set_channels: %s\n", snd_strerror(err));
181			return False;
182			}
183
184
185			buffertime = 500000; /* microseconds */
186			if ((err =
187			snd_pcm_hw_params_set_buffer_time_near(pcm_handle, hwparams, &buffertime, 0)) < 0)
188			{
189			error("snd_pcm_hw_params_set_buffer_time_near: %s\n", snd_strerror(err));
190			return False;
191			}
192
193			if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0)
194			{
195			error("snd_pcm_hw_params: %s\n", snd_strerror(err));
196			return False;
197			}
198
199			snd_pcm_hw_params_free(hwparams);
200
201			if ((err = snd_pcm_prepare(pcm_handle)) < 0)
202			{
203			error("snd_pcm_prepare: %s\n", snd_strerror(err));
204			return False;
205			}
206
207			reopened = True;
208
209			return True;
210			}
211
212			void
213			wave_out_volume(uint16 left, uint16 right)
214			{
215			static int warned = 0;
216
217			if (!warned)
218			{
219			warning("volume changes currently not supported with experimental alsa-output\n");
220			warned = 1;
221			}
222			}
223
224			void
225			wave_out_play(void)
226			{
227			struct audio_packet *packet;
228			STREAM out;
229			int len;
230			static long prev_s, prev_us;
231			unsigned int duration;
232			struct timeval tv;
233			int next_tick;
234
235			if (reopened)
236			{
237			reopened = False;
238			gettimeofday(&tv, NULL);
239			prev_s = tv.tv_sec;
240			prev_us = tv.tv_usec;
241			}
242
243	stargo	1254	if (rdpsnd_queue_empty())
244	stargo	1253	{
245			g_dsp_busy = 0;
246			return;
247			}
248
249	stargo	1254	packet = rdpsnd_queue_current_packet();
250	stargo	1253	out = &packet->s;
251
252	stargo	1254	next_tick = rdpsnd_queue_next_tick();
253	stargo	1253
254			len = (out->end - out->p) / (samplewidth * audiochannels);
255			if ((len = snd_pcm_writei(pcm_handle, out->p, ((MAX_FRAMES < len) ? MAX_FRAMES : len))) < 0)
256			{
257			snd_pcm_prepare(pcm_handle);
258			len = 0;
259			}
260			out->p += (len * samplewidth * audiochannels);
261
262			gettimeofday(&tv, NULL);
263
264			duration = ((tv.tv_sec - prev_s) * 1000000 + (tv.tv_usec - prev_us)) / 1000;
265
266			if (packet->tick > next_tick)
267			next_tick += 65536;
268
269			if ((out->p == out->end) \|\| duration > next_tick - packet->tick + 500)
270			{
271			prev_s = tv.tv_sec;
272			prev_us = tv.tv_usec;
273
274			if (abs((next_tick - packet->tick) - duration) > 20)
275			{
276			DEBUG(("duration: %d, calc: %d, ", duration, next_tick - packet->tick));
277			DEBUG(("last: %d, is: %d, should: %d\n", packet->tick,
278			(packet->tick + duration) % 65536, next_tick % 65536));
279			}
280
281	stargo	1254	rdpsnd_send_completion(((packet->tick + duration) % 65536), packet->index);
282			rdpsnd_queue_next();
283	stargo	1253	}
284
285			g_dsp_busy = 1;
286			return;
287			}