trunk/rdesktop/rdpsnd_libao.c

/* -*- c-basic-offset: 8 -*-
   rdesktop: A Remote Desktop Protocol client.
   Sound Channel Process Functions - libao-driver
   Copyright (C) Matthew Chapman 2003
   Copyright (C) GuoJunBo guojunbo@ict.ac.cn 2003
   Copyright (C) Michael Gernoth mike@zerfleddert.de 2005

   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 <unistd.h>
#include <fcntl.h>
#include <errno.h>
#include <ao/ao.h>
#include <sys/time.h>

#define MAX_QUEUE       10
#define WAVEOUTBUF      64

int g_dsp_fd;
ao_device *o_device = NULL;
int default_driver;
int g_samplerate;
int g_channels;
BOOL g_dsp_busy = False;
static BOOL g_reopened;
static short g_samplewidth;

static struct audio_packet
{
        struct stream s;
        uint16 tick;
        uint8 index;
} packet_queue[MAX_QUEUE];
static unsigned int queue_hi, queue_lo;

BOOL
wave_out_open(void)
{
        ao_sample_format format;

        ao_initialize();
        default_driver = ao_default_driver_id();

        format.bits = 16;
        format.channels = 2;
        g_channels = 2;
        format.rate = 44100;
        g_samplerate = 44100;
        format.byte_format = AO_FMT_LITTLE;

        o_device = ao_open_live(default_driver, &format, NULL);
        if (o_device == NULL)
        {
                return False;
        }

        g_dsp_fd = 0;
        queue_lo = queue_hi = 0;

        g_reopened = True;

        return True;
}

void
wave_out_close(void)
{
        /* Ack all remaining packets */
        while (queue_lo != queue_hi)
        {
                rdpsnd_send_completion(packet_queue[queue_lo].tick, packet_queue[queue_lo].index);
                free(packet_queue[queue_lo].s.data);
                queue_lo = (queue_lo + 1) % MAX_QUEUE;
        }

        if (o_device != NULL)
                ao_close(o_device);

        ao_shutdown();
}

BOOL
wave_out_format_supported(WAVEFORMATEX * pwfx)
{
        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;
        /* The only common denominator between libao output drivers is a sample-rate of
           44100, we need to upsample 22050 to it */
        if ((pwfx->nSamplesPerSec != 44100) && (pwfx->nSamplesPerSec != 22050))
                return False;

        return True;
}

BOOL
wave_out_set_format(WAVEFORMATEX * pwfx)
{
        ao_sample_format format;

        format.bits = pwfx->wBitsPerSample;
        format.channels = pwfx->nChannels;
        g_channels = pwfx->nChannels;
        format.rate = 44100;
        g_samplerate = pwfx->nSamplesPerSec;
        format.byte_format = AO_FMT_LITTLE;

        g_samplewidth = pwfx->wBitsPerSample / 8;

        if (o_device != NULL)
                ao_close(o_device);

        o_device = ao_open_live(default_driver, &format, NULL);
        if (o_device == NULL)
        {
                return False;
        }

        g_reopened = True;

        return True;
}

void
wave_out_volume(uint16 left, uint16 right)
{
        warning("volume changes not supported with libao-output\n");
}

void
wave_out_write(STREAM s, uint16 tick, uint8 index)
{
        struct audio_packet *packet = &packet_queue[queue_hi];
        unsigned int next_hi = (queue_hi + 1) % MAX_QUEUE;

        if (next_hi == queue_lo)
        {
                error("No space to queue audio packet\n");
                return;
        }

        queue_hi = next_hi;

        packet->s = *s;
        packet->tick = tick;
        packet->index = index;
        packet->s.p += 4;

        /* we steal the data buffer from s, give it a new one */
        s->data = malloc(s->size);

        if (!g_dsp_busy)
                wave_out_play();
}

void
wave_out_play(void)
{
        struct audio_packet *packet;
        STREAM out;
        unsigned char outbuf[WAVEOUTBUF];
        int offset, len, i;
        static long prev_s, prev_us;
        unsigned int duration;
        struct timeval tv;
        int next_tick;

        if (g_reopened)
        {
                g_reopened = False;
                gettimeofday(&tv, NULL);
                prev_s = tv.tv_sec;
                prev_us = tv.tv_usec;
        }

        if (queue_lo == queue_hi)
        {
                g_dsp_busy = 0;
                return;
        }

        packet = &packet_queue[queue_lo];
        out = &packet->s;

        if (((queue_lo + 1) % MAX_QUEUE) != queue_hi)
        {
                next_tick = packet_queue[(queue_lo + 1) % MAX_QUEUE].tick;
        }
        else
        {
                next_tick = (packet->tick + 65535) % 65536;
        }

        len = 0;

        if (g_samplerate == 22050)
        {
                /* Resample to 44100 */
                for (i = 0; (i < ((WAVEOUTBUF / 4) * (3 - g_samplewidth))) && (out->p < out->end);
                     i++)
                {
                        /* On a stereo-channel we must make sure that left and right
                           does not get mixed up, so we need to expand the sample-
                           data with channels in mind: 1234 -> 12123434
                           If we have a mono-channel, we can expand the data by simply
                           doubling the sample-data: 1234 -> 11223344 */
                        if (g_channels == 2)
                                offset = ((i * 2) - (i & 1)) * g_samplewidth;
                        else
                                offset = (i * 2) * g_samplewidth;

                        memcpy(&outbuf[offset], out->p, g_samplewidth);
                        memcpy(&outbuf[g_channels * g_samplewidth + offset], out->p, g_samplewidth);

                        out->p += g_samplewidth;
                        len += 2 * g_samplewidth;
                }
        }
        else
        {
                len = (WAVEOUTBUF > (out->end - out->p)) ? (out->end - out->p) : WAVEOUTBUF;
                memcpy(outbuf, out->p, len);
                out->p += len;
        }

        ao_play(o_device, (char *) outbuf, len);

        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));
                }

                /* Until all drivers are using the windows sound-ticks, we need to
                   substract the 50 ticks added later by rdpsnd.c */
                rdpsnd_send_completion(((packet->tick + duration) % 65536) - 50, packet->index);
                free(out->data);
                queue_lo = (queue_lo + 1) % MAX_QUEUE;
        }

        g_dsp_busy = 1;
        return;
}
1	/* -- c-basic-offset: 8 --
2	rdesktop: A Remote Desktop Protocol client.
3	Sound Channel Process Functions - libao-driver
4	Copyright (C) Matthew Chapman 2003
5	Copyright (C) GuoJunBo guojunbo@ict.ac.cn 2003
6	Copyright (C) Michael Gernoth mike@zerfleddert.de 2005
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	#include <unistd.h>
25	#include <fcntl.h>
26	#include <errno.h>
27	#include <ao/ao.h>
28	#include <sys/time.h>
29
30	#define MAX_QUEUE 10
31	#define WAVEOUTBUF 64
32
33	int g_dsp_fd;
34	ao_device *o_device = NULL;
35	int default_driver;
36	int g_samplerate;
37	int g_channels;
38	BOOL g_dsp_busy = False;
39	static BOOL g_reopened;
40	static short g_samplewidth;
41
42	static struct audio_packet
43	{
44	struct stream s;
45	uint16 tick;
46	uint8 index;
47	} packet_queue[MAX_QUEUE];
48	static unsigned int queue_hi, queue_lo;
49
50	BOOL
51	wave_out_open(void)
52	{
53	ao_sample_format format;
54
55	ao_initialize();
56	default_driver = ao_default_driver_id();
57
58	format.bits = 16;
59	format.channels = 2;
60	g_channels = 2;
61	format.rate = 44100;
62	g_samplerate = 44100;
63	format.byte_format = AO_FMT_LITTLE;
64
65	o_device = ao_open_live(default_driver, &format, NULL);
66	if (o_device == NULL)
67	{
68	return False;
69	}
70
71	g_dsp_fd = 0;
72	queue_lo = queue_hi = 0;
73
74	g_reopened = True;
75
76	return True;
77	}
78
79	void
80	wave_out_close(void)
81	{
82	/* Ack all remaining packets */
83	while (queue_lo != queue_hi)
84	{
85	rdpsnd_send_completion(packet_queue[queue_lo].tick, packet_queue[queue_lo].index);
86	free(packet_queue[queue_lo].s.data);
87	queue_lo = (queue_lo + 1) % MAX_QUEUE;
88	}
89
90	if (o_device != NULL)
91	ao_close(o_device);
92
93	ao_shutdown();
94	}
95
96	BOOL
97	wave_out_format_supported(WAVEFORMATEX * pwfx)
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	/* The only common denominator between libao output drivers is a sample-rate of
106	44100, we need to upsample 22050 to it */
107	if ((pwfx->nSamplesPerSec != 44100) && (pwfx->nSamplesPerSec != 22050))
108	return False;
109
110	return True;
111	}
112
113	BOOL
114	wave_out_set_format(WAVEFORMATEX * pwfx)
115	{
116	ao_sample_format format;
117
118	format.bits = pwfx->wBitsPerSample;
119	format.channels = pwfx->nChannels;
120	g_channels = pwfx->nChannels;
121	format.rate = 44100;
122	g_samplerate = pwfx->nSamplesPerSec;
123	format.byte_format = AO_FMT_LITTLE;
124
125	g_samplewidth = pwfx->wBitsPerSample / 8;
126
127	if (o_device != NULL)
128	ao_close(o_device);
129
130	o_device = ao_open_live(default_driver, &format, NULL);
131	if (o_device == NULL)
132	{
133	return False;
134	}
135
136	g_reopened = True;
137
138	return True;
139	}
140
141	void
142	wave_out_volume(uint16 left, uint16 right)
143	{
144	warning("volume changes not supported with libao-output\n");
145	}
146
147	void
148	wave_out_write(STREAM s, uint16 tick, uint8 index)
149	{
150	struct audio_packet *packet = &packet_queue[queue_hi];
151	unsigned int next_hi = (queue_hi + 1) % MAX_QUEUE;
152
153	if (next_hi == queue_lo)
154	{
155	error("No space to queue audio packet\n");
156	return;
157	}
158
159	queue_hi = next_hi;
160
161	packet->s = *s;
162	packet->tick = tick;
163	packet->index = index;
164	packet->s.p += 4;
165
166	/* we steal the data buffer from s, give it a new one */
167	s->data = malloc(s->size);
168
169	if (!g_dsp_busy)
170	wave_out_play();
171	}
172
173	void
174	wave_out_play(void)
175	{
176	struct audio_packet *packet;
177	STREAM out;
178	unsigned char outbuf[WAVEOUTBUF];
179	int offset, len, i;
180	static long prev_s, prev_us;
181	unsigned int duration;
182	struct timeval tv;
183	int next_tick;
184
185	if (g_reopened)
186	{
187	g_reopened = False;
188	gettimeofday(&tv, NULL);
189	prev_s = tv.tv_sec;
190	prev_us = tv.tv_usec;
191	}
192
193	if (queue_lo == queue_hi)
194	{
195	g_dsp_busy = 0;
196	return;
197	}
198
199	packet = &packet_queue[queue_lo];
200	out = &packet->s;
201
202	if (((queue_lo + 1) % MAX_QUEUE) != queue_hi)
203	{
204	next_tick = packet_queue[(queue_lo + 1) % MAX_QUEUE].tick;
205	}
206	else
207	{
208	next_tick = (packet->tick + 65535) % 65536;
209	}
210
211	len = 0;
212
213	if (g_samplerate == 22050)
214	{
215	/* Resample to 44100 */
216	for (i = 0; (i < ((WAVEOUTBUF / 4) * (3 - g_samplewidth))) && (out->p < out->end);
217	i++)
218	{
219	/* On a stereo-channel we must make sure that left and right
220	does not get mixed up, so we need to expand the sample-
221	data with channels in mind: 1234 -> 12123434
222	If we have a mono-channel, we can expand the data by simply
223	doubling the sample-data: 1234 -> 11223344 */
224	if (g_channels == 2)
225	offset = ((i * 2) - (i & 1)) * g_samplewidth;
226	else
227	offset = (i * 2) * g_samplewidth;
228
229	memcpy(&outbuf[offset], out->p, g_samplewidth);
230	memcpy(&outbuf[g_channels * g_samplewidth + offset], out->p, g_samplewidth);
231
232	out->p += g_samplewidth;
233	len += 2 * g_samplewidth;
234	}
235	}
236	else
237	{
238	len = (WAVEOUTBUF > (out->end - out->p)) ? (out->end - out->p) : WAVEOUTBUF;
239	memcpy(outbuf, out->p, len);
240	out->p += len;
241	}
242
243	ao_play(o_device, (char *) outbuf, len);
244
245	gettimeofday(&tv, NULL);
246
247	duration = ((tv.tv_sec - prev_s) * 1000000 + (tv.tv_usec - prev_us)) / 1000;
248
249	if (packet->tick > next_tick)
250	next_tick += 65536;
251
252	if ((out->p == out->end) \|\| duration > next_tick - packet->tick + 500)
253	{
254	prev_s = tv.tv_sec;
255	prev_us = tv.tv_usec;
256
257	if (abs((next_tick - packet->tick) - duration) > 20)
258	{
259	DEBUG(("duration: %d, calc: %d, ", duration, next_tick - packet->tick));
260	DEBUG(("last: %d, is: %d, should: %d\n", packet->tick,
261	(packet->tick + duration) % 65536, next_tick % 65536));
262	}
263
264	/* Until all drivers are using the windows sound-ticks, we need to
265	substract the 50 ticks added later by rdpsnd.c */
266	rdpsnd_send_completion(((packet->tick + duration) % 65536) - 50, packet->index);
267	free(out->data);
268	queue_lo = (queue_lo + 1) % MAX_QUEUE;
269	}
270
271	g_dsp_busy = 1;
272	return;
273	}