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;
        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, 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	astrand	963	/* -- c-basic-offset: 8 --
2	stargo	833	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	stargo	891	#include <sys/time.h>
29	stargo	833
30			#define MAX_QUEUE 10
31	stargo	840	#define WAVEOUTBUF 64
32	stargo	833
33			int g_dsp_fd;
34			ao_device *o_device = NULL;
35			int default_driver;
36	stargo	835	int g_samplerate;
37	stargo	840	int g_channels;
38	stargo	833	BOOL g_dsp_busy = False;
39	stargo	891	static BOOL g_reopened;
40	stargo	833	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	stargo	840	g_channels = 2;
61	stargo	833	format.rate = 44100;
62	stargo	835	g_samplerate = 44100;
63	stargo	833	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	stargo	891	g_reopened = True;
75
76	stargo	833	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	stargo	838
93	stargo	833	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	stargo	835	44100, we need to upsample 22050 to it */
107			if ((pwfx->nSamplesPerSec != 44100) && (pwfx->nSamplesPerSec != 22050))
108	stargo	833	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	stargo	840	g_channels = pwfx->nChannels;
121	stargo	833	format.rate = 44100;
122	stargo	835	g_samplerate = pwfx->nSamplesPerSec;
123	stargo	833	format.byte_format = AO_FMT_LITTLE;
124
125			g_samplewidth = pwfx->wBitsPerSample / 8;
126
127	stargo	837	if (o_device != NULL)
128	stargo	833	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	stargo	891	g_reopened = True;
137	stargo	833
138			return True;
139			}
140
141			void
142			wave_out_volume(uint16 left, uint16 right)
143			{
144	stargo	838	warning("volume changes not supported with libao-output\n");
145	stargo	833	}
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	stargo	1043	char outbuf[WAVEOUTBUF];
179	stargo	837	int offset, len, i;
180	stargo	891	static long prev_s, prev_us;
181			unsigned int duration;
182			struct timeval tv;
183			int next_tick;
184	stargo	833
185	stargo	891	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	stargo	835	if (queue_lo == queue_hi)
194	stargo	833	{
195	stargo	835	g_dsp_busy = 0;
196			return;
197			}
198	stargo	833
199	stargo	835	packet = &packet_queue[queue_lo];
200			out = &packet->s;
201	stargo	833
202	stargo	891	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	stargo	835	len = 0;
212	stargo	833
213	stargo	837	if (g_samplerate == 22050)
214	stargo	835	{
215			/* Resample to 44100 */
216	stargo	840	for (i = 0; (i < ((WAVEOUTBUF / 4) * (3 - g_samplewidth))) && (out->p < out->end);
217	stargo	837	i++)
218	stargo	835	{
219	stargo	841	/* 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	stargo	840	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	stargo	833
232	stargo	841	out->p += g_samplewidth;
233	stargo	840	len += 2 * g_samplewidth;
234	stargo	833	}
235			}
236	stargo	835	else
237			{
238	stargo	836	len = (WAVEOUTBUF > (out->end - out->p)) ? (out->end - out->p) : WAVEOUTBUF;
239	stargo	837	memcpy(outbuf, out->p, len);
240	stargo	835	out->p += len;
241			}
242
243	stargo	1043	ao_play(o_device, outbuf, len);
244	stargo	835
245	stargo	891	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	stargo	835	{
254	stargo	891	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	stargo	835	free(out->data);
268			queue_lo = (queue_lo + 1) % MAX_QUEUE;
269			}
270
271			g_dsp_busy = 1;
272			return;
273	stargo	833	}