reactos/modules/rosapps/applications/net/tsclient/rdesktop/rdpsnd_sgi.c

341 lines
7.7 KiB
C

/* -*- c-basic-offset: 8 -*-
rdesktop: A Remote Desktop Protocol client.
Sound Channel Process Functions - SGI/IRIX
Copyright (C) Matthew Chapman 2003
Copyright (C) GuoJunBo guojunbo@ict.ac.cn 2003
Copyright (C) Jeremy Meng void.foo@gmail.com 2004, 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.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "rdesktop.h"
#include <errno.h>
#include <dmedia/audio.h>
/* #define IRIX_DEBUG 1 */
#define IRIX_MAX_VOL 65535
#define MAX_QUEUE 10
int This->dsp_;
ALconfig audioconfig;
ALport output_port;
BOOL This->dsp_bu = False;
static BOOL g_swapaudio;
static int g_snd_rate;
static BOOL g_swapaudio;
static int width = AL_SAMPLE_16;
double min_volume, max_volume, volume_range;
int resource, maxFillable;
int combinedFrameSize;
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)
{
ALparamInfo pinfo;
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_open: begin\n");
#endif
if (alGetParamInfo(AL_DEFAULT_OUTPUT, AL_GAIN, &pinfo) < 0)
{
fprintf(stderr, "wave_out_open: alGetParamInfo failed: %s\n",
alGetErrorString(oserror()));
}
min_volume = alFixedToDouble(pinfo.min.ll);
max_volume = alFixedToDouble(pinfo.max.ll);
volume_range = (max_volume - min_volume);
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_open: minvol = %lf, maxvol= %lf, range = %lf.\n",
min_volume, max_volume, volume_range);
#endif
queue_lo = queue_hi = 0;
audioconfig = alNewConfig();
if (audioconfig == (ALconfig) 0)
{
fprintf(stderr, "wave_out_open: alNewConfig failed: %s\n",
alGetErrorString(oserror()));
return False;
}
output_port = alOpenPort("rdpsnd", "w", 0);
if (output_port == (ALport) 0)
{
fprintf(stderr, "wave_out_open: alOpenPort failed: %s\n",
alGetErrorString(oserror()));
return False;
}
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_open: returning\n");
#endif
return True;
}
void
wave_out_close(void)
{
/* Ack all remaining packets */
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_close: begin\n");
#endif
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;
}
alDiscardFrames(output_port, 0);
alClosePort(output_port);
alFreeConfig(audioconfig);
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_close: returning\n");
#endif
}
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;
return True;
}
BOOL
wave_out_set_format(WAVEFORMATEX * pwfx)
{
int channels;
int frameSize, channelCount;
ALpv params;
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_set_format: init...\n");
#endif
g_swapaudio = False;
if (pwfx->wBitsPerSample == 8)
width = AL_SAMPLE_8;
else if (pwfx->wBitsPerSample == 16)
{
width = AL_SAMPLE_16;
/* Do we need to swap the 16bit values? (Are we BigEndian) */
#if (defined(B_ENDIAN))
g_swapaudio = 1;
#else
g_swapaudio = 0;
#endif
}
/* Limited support to configure an opened audio port in IRIX. The
number of channels is a static setting and can not be changed after
a port is opened. So if the number of channels remains the same, we
can configure other settings; otherwise we have to reopen the audio
port, using same config. */
channels = pwfx->nChannels;
g_snd_rate = pwfx->nSamplesPerSec;
alSetSampFmt(audioconfig, AL_SAMPFMT_TWOSCOMP);
alSetWidth(audioconfig, width);
if (channels != alGetChannels(audioconfig))
{
alClosePort(output_port);
alSetChannels(audioconfig, channels);
output_port = alOpenPort("rdpsnd", "w", audioconfig);
if (output_port == (ALport) 0)
{
fprintf(stderr, "wave_out_set_format: alOpenPort failed: %s\n",
alGetErrorString(oserror()));
return False;
}
}
resource = alGetResource(output_port);
maxFillable = alGetFillable(output_port);
channelCount = alGetChannels(audioconfig);
frameSize = alGetWidth(audioconfig);
if (frameSize == 0 || channelCount == 0)
{
fprintf(stderr, "wave_out_set_format: bad frameSize or channelCount\n");
return False;
}
combinedFrameSize = frameSize * channelCount;
params.param = AL_RATE;
params.value.ll = (long long) g_snd_rate << 32;
if (alSetParams(resource, &params, 1) < 0)
{
fprintf(stderr, "wave_set_format: alSetParams failed: %s\n",
alGetErrorString(oserror()));
return False;
}
if (params.sizeOut < 0)
{
fprintf(stderr, "wave_set_format: invalid rate %d\n", g_snd_rate);
return False;
}
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_set_format: returning...\n");
#endif
return True;
}
void
wave_out_volume(uint16 left, uint16 right)
{
double gainleft, gainright;
ALpv pv[1];
ALfixed gain[8];
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_volume: begin\n");
fprintf(stderr, "left='%d', right='%d'\n", left, right);
#endif
gainleft = (double) left / IRIX_MAX_VOL;
gainright = (double) right / IRIX_MAX_VOL;
gain[0] = alDoubleToFixed(min_volume + gainleft * volume_range);
gain[1] = alDoubleToFixed(min_volume + gainright * volume_range);
pv[0].param = AL_GAIN;
pv[0].value.ptr = gain;
pv[0].sizeIn = 8;
if (alSetParams(AL_DEFAULT_OUTPUT, pv, 1) < 0)
{
fprintf(stderr, "wave_out_volume: alSetParams failed: %s\n",
alGetErrorString(oserror()));
return;
}
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_volume: returning\n");
#endif
}
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)
{
fprintf(stderr, "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 (!This->dsp_bu)
wave_out_play();
}
void
wave_out_play(void)
{
struct audio_packet *packet;
ssize_t len;
unsigned int i;
uint8 swap;
STREAM out;
static BOOL swapped = False;
int gf;
while (1)
{
if (queue_lo == queue_hi)
{
This->dsp_bu = False;
return;
}
packet = &packet_queue[queue_lo];
out = &packet->s;
/* Swap the current packet, but only once */
if (g_swapaudio && !swapped)
{
for (i = 0; i < out->end - out->p; i += 2)
{
swap = *(out->p + i);
*(out->p + i) = *(out->p + i + 1);
*(out->p + i + 1) = swap;
}
swapped = True;
}
len = out->end - out->p;
alWriteFrames(output_port, out->p, len / combinedFrameSize);
out->p += len;
if (out->p == out->end)
{
gf = alGetFilled(output_port);
if (gf < (4 * maxFillable / 10))
{
rdpsnd_send_completion(packet->tick, packet->index);
free(out->data);
queue_lo = (queue_lo + 1) % MAX_QUEUE;
swapped = False;
}
else
{
#if (defined(IRIX_DEBUG))
/* fprintf(stderr,"Busy playing...\n"); */
#endif
This->dsp_bu = True;
usleep(10);
return;
}
}
}
}