trunk/rdesktop/rdpsnd_libao.c

/* 
   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>

#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 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;

        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;
        }


        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;

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

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

        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);

        if (out->p == out->end)
        {
                rdpsnd_send_completion(packet->tick, packet->index);
                free(out->data);
                queue_lo = (queue_lo + 1) % MAX_QUEUE;
        }

        g_dsp_busy = 1;
        return;
}
1	/*
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
29	#define MAX_QUEUE 10
30	#define WAVEOUTBUF 64
31
32	int g_dsp_fd;
33	ao_device *o_device = NULL;
34	int default_driver;
35	int g_samplerate;
36	int g_channels;
37	BOOL g_dsp_busy = False;
38	static short g_samplewidth;
39
40	static struct audio_packet
41	{
42	struct stream s;
43	uint16 tick;
44	uint8 index;
45	} packet_queue[MAX_QUEUE];
46	static unsigned int queue_hi, queue_lo;
47
48	BOOL
49	wave_out_open(void)
50	{
51	ao_sample_format format;
52
53	ao_initialize();
54	default_driver = ao_default_driver_id();
55
56	format.bits = 16;
57	format.channels = 2;
58	g_channels = 2;
59	format.rate = 44100;
60	g_samplerate = 44100;
61	format.byte_format = AO_FMT_LITTLE;
62
63	o_device = ao_open_live(default_driver, &format, NULL);
64	if (o_device == NULL)
65	{
66	return False;
67	}
68
69	g_dsp_fd = 0;
70	queue_lo = queue_hi = 0;
71
72	return True;
73	}
74
75	void
76	wave_out_close(void)
77	{
78	/* Ack all remaining packets */
79	while (queue_lo != queue_hi)
80	{
81	rdpsnd_send_completion(packet_queue[queue_lo].tick, packet_queue[queue_lo].index);
82	free(packet_queue[queue_lo].s.data);
83	queue_lo = (queue_lo + 1) % MAX_QUEUE;
84	}
85
86	if (o_device != NULL)
87	ao_close(o_device);
88
89	ao_shutdown();
90	}
91
92	BOOL
93	wave_out_format_supported(WAVEFORMATEX * pwfx)
94	{
95	if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
96	return False;
97	if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
98	return False;
99	if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
100	return False;
101	/* The only common denominator between libao output drivers is a sample-rate of
102	44100, we need to upsample 22050 to it */
103	if ((pwfx->nSamplesPerSec != 44100) && (pwfx->nSamplesPerSec != 22050))
104	return False;
105
106	return True;
107	}
108
109	BOOL
110	wave_out_set_format(WAVEFORMATEX * pwfx)
111	{
112	ao_sample_format format;
113
114	format.bits = pwfx->wBitsPerSample;
115	format.channels = pwfx->nChannels;
116	g_channels = pwfx->nChannels;
117	format.rate = 44100;
118	g_samplerate = pwfx->nSamplesPerSec;
119	format.byte_format = AO_FMT_LITTLE;
120
121	g_samplewidth = pwfx->wBitsPerSample / 8;
122
123	if (o_device != NULL)
124	ao_close(o_device);
125
126	o_device = ao_open_live(default_driver, &format, NULL);
127	if (o_device == NULL)
128	{
129	return False;
130	}
131
132
133	return True;
134	}
135
136	void
137	wave_out_volume(uint16 left, uint16 right)
138	{
139	warning("volume changes not supported with libao-output\n");
140	}
141
142	void
143	wave_out_write(STREAM s, uint16 tick, uint8 index)
144	{
145	struct audio_packet *packet = &packet_queue[queue_hi];
146	unsigned int next_hi = (queue_hi + 1) % MAX_QUEUE;
147
148	if (next_hi == queue_lo)
149	{
150	error("No space to queue audio packet\n");
151	return;
152	}
153
154	queue_hi = next_hi;
155
156	packet->s = *s;
157	packet->tick = tick;
158	packet->index = index;
159	packet->s.p += 4;
160
161	/* we steal the data buffer from s, give it a new one */
162	s->data = malloc(s->size);
163
164	if (!g_dsp_busy)
165	wave_out_play();
166	}
167
168	void
169	wave_out_play(void)
170	{
171	struct audio_packet *packet;
172	STREAM out;
173	unsigned char outbuf[WAVEOUTBUF];
174	int offset, len, i;
175
176	if (queue_lo == queue_hi)
177	{
178	g_dsp_busy = 0;
179	return;
180	}
181
182	packet = &packet_queue[queue_lo];
183	out = &packet->s;
184
185	len = 0;
186
187	if (g_samplerate == 22050)
188	{
189	/* Resample to 44100 */
190	for (i = 0; (i < ((WAVEOUTBUF / 4) * (3 - g_samplewidth))) && (out->p < out->end);
191	i++)
192	{
193	/* On a stereo-channel we must make sure that left and right
194	does not get mixed up, so we need to expand the sample-
195	data with channels in mind: 1234 -> 12123434
196	If we have a mono-channel, we can expand the data by simply
197	doubling the sample-data: 1234 -> 11223344 */
198	if (g_channels == 2)
199	offset = ((i * 2) - (i & 1)) * g_samplewidth;
200	else
201	offset = (i * 2) * g_samplewidth;
202
203	memcpy(&outbuf[offset], out->p, g_samplewidth);
204	memcpy(&outbuf[g_channels * g_samplewidth + offset], out->p, g_samplewidth);
205
206	out->p += g_samplewidth;
207	len += 2 * g_samplewidth;
208	}
209	}
210	else
211	{
212	len = (WAVEOUTBUF > (out->end - out->p)) ? (out->end - out->p) : WAVEOUTBUF;
213	memcpy(outbuf, out->p, len);
214	out->p += len;
215	}
216
217	ao_play(o_device, outbuf, len);
218
219	if (out->p == out->end)
220	{
221	rdpsnd_send_completion(packet->tick, packet->index);
222	free(out->data);
223	queue_lo = (queue_lo + 1) % MAX_QUEUE;
224	}
225
226	g_dsp_busy = 1;
227	return;
228	}