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 2003-2008
   Copyright (C) GuoJunBo guojunbo@ict.ac.cn 2003
   Copyright (C) Michael Gernoth mike@zerfleddert.de 2006-2008
   Copyright 2006-2008 Pierre Ossman <ossman@cendio.se> for Cendio AB

   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 struct pollfd pfds_out[32];
static int num_fds_out;

static struct pollfd pfds_in[32];
static int num_fds_in;

static snd_pcm_t *out_handle = NULL;
static snd_pcm_t *in_handle = NULL;

static RD_BOOL reopened;

static short samplewidth_out;
static int audiochannels_out;
static unsigned int rate_out;

static short samplewidth_in;
static int audiochannels_in;
static unsigned int rate_in;

static char *pcm_name;

void alsa_play(void);
void alsa_record(void);

void
alsa_add_fds(int *n, fd_set * rfds, fd_set * wfds, struct timeval *tv)
{
        int err;
        struct pollfd *f;

        if (out_handle && !rdpsnd_queue_empty())
        {
                num_fds_out = snd_pcm_poll_descriptors_count(out_handle);

                if (num_fds_out > sizeof(pfds_out) / sizeof(*pfds_out))
                        return;

                err = snd_pcm_poll_descriptors(out_handle, pfds_out, num_fds_out);
                if (err < 0)
                        return;

                for (f = pfds_out; f < &pfds_out[num_fds_out]; f++)
                {
                        if (f->events & POLLIN)
                                FD_SET(f->fd, rfds);
                        if (f->events & POLLOUT)
                                FD_SET(f->fd, wfds);
                        if (f->fd > *n && (f->events & (POLLIN | POLLOUT)))
                                *n = f->fd;
                }
        }

        if (in_handle)
        {
                num_fds_in = snd_pcm_poll_descriptors_count(in_handle);

                if (num_fds_in > sizeof(pfds_in) / sizeof(*pfds_in))
                        return;

                err = snd_pcm_poll_descriptors(in_handle, pfds_in, num_fds_in);
                if (err < 0)
                        return;

                for (f = pfds_in; f < &pfds_in[num_fds_in]; f++)
                {
                        if (f->events & POLLIN)
                                FD_SET(f->fd, rfds);
                        if (f->events & POLLOUT)
                                FD_SET(f->fd, wfds);
                        if (f->fd > *n && (f->events & (POLLIN | POLLOUT)))
                                *n = f->fd;
                }
        }
}

void
alsa_check_fds(fd_set * rfds, fd_set * wfds)
{
        struct pollfd *f;
        int err;
        unsigned short revents;

        if (out_handle && !rdpsnd_queue_empty())
        {
                for (f = pfds_out; f < &pfds_out[num_fds_out]; f++)
                {
                        f->revents = 0;
                        if (f->fd != -1)
                        {
                                /* Fixme: This doesn't properly deal with things like POLLHUP */
                                if (FD_ISSET(f->fd, rfds))
                                        f->revents |= POLLIN;
                                if (FD_ISSET(f->fd, wfds))
                                        f->revents |= POLLOUT;
                        }
                }

                err = snd_pcm_poll_descriptors_revents(out_handle, pfds_out, num_fds_out, &revents);
                if (err < 0)
                        return;

                if (revents & POLLOUT)
                        alsa_play();
        }


        if (in_handle)
        {
                for (f = pfds_in; f < &pfds_in[num_fds_in]; f++)
                {
                        f->revents = 0;
                        if (f->fd != -1)
                        {
                                /* Fixme: This doesn't properly deal with things like POLLHUP */
                                if (FD_ISSET(f->fd, rfds))
                                        f->revents |= POLLIN;
                                if (FD_ISSET(f->fd, wfds))
                                        f->revents |= POLLOUT;
                        }
                }

                err = snd_pcm_poll_descriptors_revents(in_handle, pfds_in, num_fds_in, &revents);
                if (err < 0)
                        return;

                if (revents & POLLIN)
                        alsa_record();
        }
}

static RD_BOOL
alsa_set_format(snd_pcm_t * pcm, RD_WAVEFORMATEX * pwfx)
{
        snd_pcm_hw_params_t *hwparams = NULL;
        int err;
        unsigned int buffertime;
        short samplewidth;
        int audiochannels;
        unsigned int rate;

        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, hwparams)) < 0)
        {
                error("snd_pcm_hw_params_any: %s\n", snd_strerror(err));
                return False;
        }

        if ((err = snd_pcm_hw_params_set_access(pcm, 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, 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, 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, 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, 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, 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, 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, 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)) < 0)
        {
                error("snd_pcm_prepare: %s\n", snd_strerror(err));
                return False;
        }

        reopened = True;

        return True;
}

RD_BOOL
alsa_open_out(void)
{
        int err;

        if ((err = snd_pcm_open(&out_handle, pcm_name, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
        {
                error("snd_pcm_open: %s\n", snd_strerror(err));
                return False;
        }

        reopened = True;

        return True;
}

void
alsa_close_out(void)
{
        /* Ack all remaining packets */
        while (!rdpsnd_queue_empty())
                rdpsnd_queue_next(0);

        if (out_handle)
        {
                snd_pcm_close(out_handle);
                out_handle = NULL;
        }
}

RD_BOOL
alsa_format_supported(RD_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;
}

RD_BOOL
alsa_set_format_out(RD_WAVEFORMATEX * pwfx)
{
        if (!alsa_set_format(out_handle, pwfx))
                return False;

        samplewidth_out = pwfx->wBitsPerSample / 8;
        audiochannels_out = pwfx->nChannels;
        rate_out = pwfx->nSamplesPerSec;

        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;
        }

        /* We shouldn't be called if the queue is empty, but still */
        if (rdpsnd_queue_empty())
                return;

        packet = rdpsnd_queue_current_packet();
        out = &packet->s;

        next_tick = rdpsnd_queue_next_tick();

        len = (out->end - out->p) / (samplewidth_out * audiochannels_out);
        if ((len = snd_pcm_writei(out_handle, out->p, ((MAX_FRAMES < len) ? MAX_FRAMES : len))) < 0)
        {
                printf("Fooo!\n");
                snd_pcm_prepare(out_handle);
                len = 0;
        }
        out->p += (len * samplewidth_out * audiochannels_out);

        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(out_handle, &delay_frames) < 0)
                        delay_frames = out->size / (samplewidth_out * audiochannels_out);
                if (delay_frames < 0)
                        delay_frames = 0;

                delay_us = delay_frames * (1000000 / rate_out);

                rdpsnd_queue_next(delay_us);
        }
}

RD_BOOL
alsa_open_in(void)
{
        int err;

        if ((err =
             snd_pcm_open(&in_handle, pcm_name, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK)) < 0)
        {
                error("snd_pcm_open: %s\n", snd_strerror(err));
                return False;
        }

        return True;
}

void
alsa_close_in(void)
{
        if (in_handle)
        {
                snd_pcm_close(in_handle);
                in_handle = NULL;
        }
}

RD_BOOL
alsa_set_format_in(RD_WAVEFORMATEX * pwfx)
{
        int err;

        if (!alsa_set_format(in_handle, pwfx))
                return False;

        if ((err = snd_pcm_start(in_handle)) < 0)
        {
                error("snd_pcm_start: %s\n", snd_strerror(err));
                return False;
        }

        samplewidth_in = pwfx->wBitsPerSample / 8;
        audiochannels_in = pwfx->nChannels;
        rate_in = pwfx->nSamplesPerSec;

        return True;
}

void
alsa_record(void)
{
        int len;
        char buffer[32768];

        len = snd_pcm_readi(in_handle, buffer,
                            sizeof(buffer) / (samplewidth_in * audiochannels_in));
        if (len < 0)
        {
                snd_pcm_prepare(in_handle);
                len = 0;
        }

        rdpsnd_record(buffer, len * samplewidth_in * audiochannels_in);
}

struct audio_driver *
alsa_register(char *options)
{
        static struct audio_driver alsa_driver;

        memset(&alsa_driver, 0, sizeof(alsa_driver));

        alsa_driver.name = "alsa";
        alsa_driver.description = "ALSA output driver, default device: " DEFAULTDEVICE;

        alsa_driver.add_fds = alsa_add_fds;
        alsa_driver.check_fds = alsa_check_fds;

        alsa_driver.wave_out_open = alsa_open_out;
        alsa_driver.wave_out_close = alsa_close_out;
        alsa_driver.wave_out_format_supported = alsa_format_supported;
        alsa_driver.wave_out_set_format = alsa_set_format_out;
        alsa_driver.wave_out_volume = rdpsnd_dsp_softvol_set;

        alsa_driver.wave_in_open = alsa_open_in;
        alsa_driver.wave_in_close = alsa_close_in;
        alsa_driver.wave_in_format_supported = alsa_format_supported;
        alsa_driver.wave_in_set_format = alsa_set_format_in;
        alsa_driver.wave_in_volume = NULL;      /* FIXME */

        alsa_driver.need_byteswap_on_be = 0;
        alsa_driver.need_resampling = 0;

        if (options)
        {
                pcm_name = xstrdup(options);
        }
        else
        {
                pcm_name = 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	jsorg71	1475	Copyright (C) Matthew Chapman 2003-2008
5	stargo	1253	Copyright (C) GuoJunBo guojunbo@ict.ac.cn 2003
6	jsorg71	1475	Copyright (C) Michael Gernoth mike@zerfleddert.de 2006-2008
7			Copyright 2006-2008 Pierre Ossman <ossman@cendio.se> for Cendio AB
8	stargo	1253
9			This program is free software; you can redistribute it and/or modify
10			it under the terms of the GNU General Public License as published by
11			the Free Software Foundation; either version 2 of the License, or
12			(at your option) any later version.
13
14			This program is distributed in the hope that it will be useful,
15			but WITHOUT ANY WARRANTY; without even the implied warranty of
16			MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17			GNU General Public License for more details.
18
19			You should have received a copy of the GNU General Public License
20			along with this program; if not, write to the Free Software
21			Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
22			*/
23
24			#include "rdesktop.h"
25	stargo	1254	#include "rdpsnd.h"
26	stargo	1261	#include "rdpsnd_dsp.h"
27	stargo	1253	#include <unistd.h>
28			#include <fcntl.h>
29			#include <errno.h>
30			#include <alsa/asoundlib.h>
31			#include <sys/time.h>
32
33			#define DEFAULTDEVICE "default"
34			#define MAX_FRAMES 32
35
36	ossman_	1360	static struct pollfd pfds_out[32];
37			static int num_fds_out;
38	ossman_	1346
39	ossman_	1360	static struct pollfd pfds_in[32];
40			static int num_fds_in;
41
42			static snd_pcm_t *out_handle = NULL;
43			static snd_pcm_t *in_handle = NULL;
44
45	ossman_	1375	static RD_BOOL reopened;
46	ossman_	1360
47			static short samplewidth_out;
48			static int audiochannels_out;
49			static unsigned int rate_out;
50
51			static short samplewidth_in;
52			static int audiochannels_in;
53			static unsigned int rate_in;
54
55	stargo	1255	static char *pcm_name;
56	stargo	1253
57	ossman_	1346	void alsa_play(void);
58	ossman_	1360	void alsa_record(void);
59	ossman_	1346
60			void
61			alsa_add_fds(int n, fd_set rfds, fd_set * wfds, struct timeval *tv)
62			{
63			int err;
64			struct pollfd *f;
65
66	ossman_	1360	if (out_handle && !rdpsnd_queue_empty())
67			{
68			num_fds_out = snd_pcm_poll_descriptors_count(out_handle);
69	ossman_	1346
70	ossman_	1360	if (num_fds_out > sizeof(pfds_out) / sizeof(*pfds_out))
71			return;
72	ossman_	1346
73	ossman_	1360	err = snd_pcm_poll_descriptors(out_handle, pfds_out, num_fds_out);
74			if (err < 0)
75			return;
76	ossman_	1346
77	ossman_	1360	for (f = pfds_out; f < &pfds_out[num_fds_out]; f++)
78			{
79			if (f->events & POLLIN)
80			FD_SET(f->fd, rfds);
81			if (f->events & POLLOUT)
82			FD_SET(f->fd, wfds);
83			if (f->fd > *n && (f->events & (POLLIN \| POLLOUT)))
84			*n = f->fd;
85			}
86			}
87	ossman_	1346
88	ossman_	1360	if (in_handle)
89			{
90			num_fds_in = snd_pcm_poll_descriptors_count(in_handle);
91	ossman_	1346
92	ossman_	1360	if (num_fds_in > sizeof(pfds_in) / sizeof(*pfds_in))
93			return;
94
95			err = snd_pcm_poll_descriptors(in_handle, pfds_in, num_fds_in);
96			if (err < 0)
97			return;
98
99			for (f = pfds_in; f < &pfds_in[num_fds_in]; f++)
100			{
101			if (f->events & POLLIN)
102			FD_SET(f->fd, rfds);
103			if (f->events & POLLOUT)
104			FD_SET(f->fd, wfds);
105			if (f->fd > *n && (f->events & (POLLIN \| POLLOUT)))
106			*n = f->fd;
107			}
108	ossman_	1346	}
109			}
110
111			void
112			alsa_check_fds(fd_set * rfds, fd_set * wfds)
113			{
114			struct pollfd *f;
115			int err;
116			unsigned short revents;
117
118	ossman_	1360	if (out_handle && !rdpsnd_queue_empty())
119	ossman_	1346	{
120	ossman_	1360	for (f = pfds_out; f < &pfds_out[num_fds_out]; f++)
121	ossman_	1346	{
122	ossman_	1360	f->revents = 0;
123			if (f->fd != -1)
124			{
125			/* Fixme: This doesn't properly deal with things like POLLHUP */
126			if (FD_ISSET(f->fd, rfds))
127			f->revents \|= POLLIN;
128			if (FD_ISSET(f->fd, wfds))
129			f->revents \|= POLLOUT;
130			}
131	ossman_	1346	}
132
133	ossman_	1360	err = snd_pcm_poll_descriptors_revents(out_handle, pfds_out, num_fds_out, &revents);
134			if (err < 0)
135			return;
136	ossman_	1346
137	ossman_	1360	if (revents & POLLOUT)
138			alsa_play();
139	stargo	1253	}
140
141
142	ossman_	1360	if (in_handle)
143	stargo	1253	{
144	ossman_	1360	for (f = pfds_in; f < &pfds_in[num_fds_in]; f++)
145			{
146			f->revents = 0;
147			if (f->fd != -1)
148			{
149			/* Fixme: This doesn't properly deal with things like POLLHUP */
150			if (FD_ISSET(f->fd, rfds))
151			f->revents \|= POLLIN;
152			if (FD_ISSET(f->fd, wfds))
153			f->revents \|= POLLOUT;
154			}
155			}
156	stargo	1253
157	ossman_	1360	err = snd_pcm_poll_descriptors_revents(in_handle, pfds_in, num_fds_in, &revents);
158			if (err < 0)
159			return;
160	stargo	1253
161	ossman_	1360	if (revents & POLLIN)
162			alsa_record();
163	stargo	1253	}
164			}
165
166	ossman_	1375	static RD_BOOL
167	jsorg71	1364	alsa_set_format(snd_pcm_t * pcm, RD_WAVEFORMATEX * pwfx)
168	stargo	1253	{
169			snd_pcm_hw_params_t *hwparams = NULL;
170			int err;
171			unsigned int buffertime;
172	ossman_	1360	short samplewidth;
173			int audiochannels;
174			unsigned int rate;
175	stargo	1253
176			samplewidth = pwfx->wBitsPerSample / 8;
177
178			if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
179			{
180			error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
181			return False;
182			}
183
184	ossman_	1360	if ((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
185	stargo	1253	{
186			error("snd_pcm_hw_params_any: %s\n", snd_strerror(err));
187			return False;
188			}
189
190	ossman_	1360	if ((err = snd_pcm_hw_params_set_access(pcm, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
191	stargo	1253	{
192			error("snd_pcm_hw_params_set_access: %s\n", snd_strerror(err));
193			return False;
194			}
195
196			if (pwfx->wBitsPerSample == 16)
197			{
198	ossman_	1360	if ((err = snd_pcm_hw_params_set_format(pcm, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
199	stargo	1253	{
200			error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
201			return False;
202			}
203			}
204			else
205			{
206	ossman_	1360	if ((err = snd_pcm_hw_params_set_format(pcm, hwparams, SND_PCM_FORMAT_S8)) < 0)
207	stargo	1253	{
208			error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
209			return False;
210			}
211			}
212
213			#if 0
214	ossman_	1360	if ((err = snd_pcm_hw_params_set_rate_resample(pcm, hwparams, 1)) < 0)
215	stargo	1253	{
216			error("snd_pcm_hw_params_set_rate_resample: %s\n", snd_strerror(err));
217			return False;
218			}
219			#endif
220
221	ossman_	1302	rate = pwfx->nSamplesPerSec;
222	ossman_	1360	if ((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0)
223	stargo	1253	{
224			error("snd_pcm_hw_params_set_rate_near: %s\n", snd_strerror(err));
225			return False;
226			}
227
228			audiochannels = pwfx->nChannels;
229	ossman_	1360	if ((err = snd_pcm_hw_params_set_channels(pcm, hwparams, pwfx->nChannels)) < 0)
230	stargo	1253	{
231			error("snd_pcm_hw_params_set_channels: %s\n", snd_strerror(err));
232			return False;
233			}
234
235
236			buffertime = 500000; /* microseconds */
237	ossman_	1360	if ((err = snd_pcm_hw_params_set_buffer_time_near(pcm, hwparams, &buffertime, 0)) < 0)
238	stargo	1253	{
239			error("snd_pcm_hw_params_set_buffer_time_near: %s\n", snd_strerror(err));
240			return False;
241			}
242
243	ossman_	1360	if ((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
244	stargo	1253	{
245			error("snd_pcm_hw_params: %s\n", snd_strerror(err));
246			return False;
247			}
248
249			snd_pcm_hw_params_free(hwparams);
250
251	ossman_	1360	if ((err = snd_pcm_prepare(pcm)) < 0)
252	stargo	1253	{
253			error("snd_pcm_prepare: %s\n", snd_strerror(err));
254			return False;
255			}
256
257			reopened = True;
258
259			return True;
260			}
261
262	ossman_	1375	RD_BOOL
263	ossman_	1360	alsa_open_out(void)
264			{
265			int err;
266
267			if ((err = snd_pcm_open(&out_handle, pcm_name, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
268			{
269			error("snd_pcm_open: %s\n", snd_strerror(err));
270			return False;
271			}
272
273			reopened = True;
274
275			return True;
276			}
277
278	stargo	1253	void
279	ossman_	1360	alsa_close_out(void)
280			{
281			/* Ack all remaining packets */
282			while (!rdpsnd_queue_empty())
283			rdpsnd_queue_next(0);
284
285			if (out_handle)
286			{
287			snd_pcm_close(out_handle);
288			out_handle = NULL;
289			}
290			}
291
292	ossman_	1375	RD_BOOL
293	stargo	1369	alsa_format_supported(RD_WAVEFORMATEX * pwfx)
294	ossman_	1360	{
295			#if 0
296			int err;
297			snd_pcm_hw_params_t *hwparams = NULL;
298
299			if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
300			{
301			error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
302			return False;
303			}
304
305			if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
306			{
307			error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
308			return False;
309			}
310			snd_pcm_hw_params_free(hwparams);
311			#endif
312
313			if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
314			return False;
315			if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
316			return False;
317			if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
318			return False;
319			if ((pwfx->nSamplesPerSec != 44100) && (pwfx->nSamplesPerSec != 22050))
320			return False;
321
322			return True;
323			}
324
325	ossman_	1375	RD_BOOL
326	stargo	1369	alsa_set_format_out(RD_WAVEFORMATEX * pwfx)
327	ossman_	1360	{
328			if (!alsa_set_format(out_handle, pwfx))
329			return False;
330
331			samplewidth_out = pwfx->wBitsPerSample / 8;
332			audiochannels_out = pwfx->nChannels;
333			rate_out = pwfx->nSamplesPerSec;
334
335			return True;
336			}
337
338			void
339	stargo	1255	alsa_play(void)
340	stargo	1253	{
341			struct audio_packet *packet;
342			STREAM out;
343			int len;
344			static long prev_s, prev_us;
345			unsigned int duration;
346			struct timeval tv;
347			int next_tick;
348
349			if (reopened)
350			{
351			reopened = False;
352			gettimeofday(&tv, NULL);
353			prev_s = tv.tv_sec;
354			prev_us = tv.tv_usec;
355			}
356
357	ossman_	1346	/* We shouldn't be called if the queue is empty, but still */
358	stargo	1254	if (rdpsnd_queue_empty())
359	stargo	1253	return;
360
361	stargo	1254	packet = rdpsnd_queue_current_packet();
362	stargo	1253	out = &packet->s;
363
364	stargo	1254	next_tick = rdpsnd_queue_next_tick();
365	stargo	1253
366	ossman_	1360	len = (out->end - out->p) / (samplewidth_out * audiochannels_out);
367			if ((len = snd_pcm_writei(out_handle, out->p, ((MAX_FRAMES < len) ? MAX_FRAMES : len))) < 0)
368	stargo	1253	{
369	ossman_	1360	printf("Fooo!\n");
370			snd_pcm_prepare(out_handle);
371	stargo	1253	len = 0;
372			}
373	ossman_	1360	out->p += (len * samplewidth_out * audiochannels_out);
374	stargo	1253
375			gettimeofday(&tv, NULL);
376
377			duration = ((tv.tv_sec - prev_s) * 1000000 + (tv.tv_usec - prev_us)) / 1000;
378
379			if (packet->tick > next_tick)
380			next_tick += 65536;
381
382			if ((out->p == out->end) \|\| duration > next_tick - packet->tick + 500)
383			{
384	ossman_	1302	snd_pcm_sframes_t delay_frames;
385			unsigned long delay_us;
386
387	stargo	1253	prev_s = tv.tv_sec;
388			prev_us = tv.tv_usec;
389
390			if (abs((next_tick - packet->tick) - duration) > 20)
391			{
392			DEBUG(("duration: %d, calc: %d, ", duration, next_tick - packet->tick));
393			DEBUG(("last: %d, is: %d, should: %d\n", packet->tick,
394			(packet->tick + duration) % 65536, next_tick % 65536));
395			}
396
397	ossman_	1360	if (snd_pcm_delay(out_handle, &delay_frames) < 0)
398			delay_frames = out->size / (samplewidth_out * audiochannels_out);
399	ossman_	1302	if (delay_frames < 0)
400			delay_frames = 0;
401
402	ossman_	1360	delay_us = delay_frames * (1000000 / rate_out);
403	ossman_	1302
404			rdpsnd_queue_next(delay_us);
405	stargo	1253	}
406			}
407	stargo	1255
408	ossman_	1375	RD_BOOL
409	ossman_	1360	alsa_open_in(void)
410			{
411			int err;
412
413			if ((err =
414			snd_pcm_open(&in_handle, pcm_name, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK)) < 0)
415			{
416			error("snd_pcm_open: %s\n", snd_strerror(err));
417			return False;
418			}
419
420			return True;
421			}
422
423			void
424			alsa_close_in(void)
425			{
426			if (in_handle)
427			{
428			snd_pcm_close(in_handle);
429			in_handle = NULL;
430			}
431			}
432
433	ossman_	1375	RD_BOOL
434	stargo	1369	alsa_set_format_in(RD_WAVEFORMATEX * pwfx)
435	ossman_	1360	{
436			int err;
437
438			if (!alsa_set_format(in_handle, pwfx))
439			return False;
440
441			if ((err = snd_pcm_start(in_handle)) < 0)
442			{
443			error("snd_pcm_start: %s\n", snd_strerror(err));
444			return False;
445			}
446
447			samplewidth_in = pwfx->wBitsPerSample / 8;
448			audiochannels_in = pwfx->nChannels;
449			rate_in = pwfx->nSamplesPerSec;
450
451			return True;
452			}
453
454			void
455			alsa_record(void)
456			{
457			int len;
458			char buffer[32768];
459
460			len = snd_pcm_readi(in_handle, buffer,
461			sizeof(buffer) / (samplewidth_in * audiochannels_in));
462			if (len < 0)
463			{
464			snd_pcm_prepare(in_handle);
465			len = 0;
466			}
467
468			rdpsnd_record(buffer, len * samplewidth_in * audiochannels_in);
469			}
470
471	stargo	1351	struct audio_driver *
472			alsa_register(char *options)
473			{
474			static struct audio_driver alsa_driver;
475	ossman_	1345
476	ossman_	1356	memset(&alsa_driver, 0, sizeof(alsa_driver));
477	ossman_	1346
478	ossman_	1356	alsa_driver.name = "alsa";
479			alsa_driver.description = "ALSA output driver, default device: " DEFAULTDEVICE;
480
481	stargo	1351	alsa_driver.add_fds = alsa_add_fds;
482			alsa_driver.check_fds = alsa_check_fds;
483	ossman_	1345
484	ossman_	1360	alsa_driver.wave_out_open = alsa_open_out;
485			alsa_driver.wave_out_close = alsa_close_out;
486	stargo	1351	alsa_driver.wave_out_format_supported = alsa_format_supported;
487	ossman_	1360	alsa_driver.wave_out_set_format = alsa_set_format_out;
488	stargo	1351	alsa_driver.wave_out_volume = rdpsnd_dsp_softvol_set;
489	ossman_	1345
490	ossman_	1360	alsa_driver.wave_in_open = alsa_open_in;
491			alsa_driver.wave_in_close = alsa_close_in;
492			alsa_driver.wave_in_format_supported = alsa_format_supported;
493			alsa_driver.wave_in_set_format = alsa_set_format_in;
494			alsa_driver.wave_in_volume = NULL; /* FIXME */
495
496	stargo	1351	alsa_driver.need_byteswap_on_be = 0;
497			alsa_driver.need_resampling = 0;
498
499	stargo	1255	if (options)
500			{
501			pcm_name = xstrdup(options);
502			}
503			else
504			{
505	ossman_	1356	pcm_name = DEFAULTDEVICE;
506	stargo	1255	}
507
508			return &alsa_driver;
509			}