SGI sound support from Jeremy Meng <voidfoo@cwazy.co.uk>
git-svn-id: svn://svn.code.sf.net/p/rdesktop/code/trunk/rdesktop@743 423420c4-83ab-492f-b58f-81f9feb106b5
This commit is contained in:
parent
8c76c35458
commit
d5cdeaa8de
13
configure.ac
13
configure.ac
@ -132,12 +132,13 @@ AC_ARG_WITH(libvncserver,
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#
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sound="yes"
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AC_ARG_WITH(sound,
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[ --with-sound select sound system ("oss" or "sun") ],
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[ --with-sound select sound system ("oss", "sgi" or "sun") ],
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[
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sound="$withval"
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])
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if test "$sound" = yes; then
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AC_CHECK_HEADER(sys/soundcard.h, [sound=oss])
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AC_CHECK_HEADER(dmedia/audio.h, [sound=sgi])
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AC_CHECK_HEADER(sys/audioio.h, [sound=sun])
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fi
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if test "$sound" = no; then
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@ -145,12 +146,16 @@ if test "$sound" = no; then
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elif test "$sound" = oss; then
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SOUNDOBJ="rdpsnd.o rdpsnd_oss.o"
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AC_DEFINE(WITH_RDPSND)
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elif test "$sound" = sgi; then
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SOUNDOBJ="rdpsnd.o rdpsnd_sgi.o"
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LDFLAGS="$LDFLAGS -laudio"
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AC_DEFINE(WITH_RDPSND)
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elif test "$sound" = sun; then
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SOUNDOBJ="rdpsnd.o rdpsnd_sun.o"
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AC_DEFINE(WITH_RDPSND)
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else
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AC_MSG_WARN([sound support disabled (no sys/soundcard.h or sys/audioio.h)])
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AC_MSG_WARN([Currently supported systems are Open Sound System (oss) and Sun/BSD (sun)])
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AC_MSG_WARN([Currently supported systems are Open Sound System (oss), SGI AL (sgi) and Sun/BSD (sun)])
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fi
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AC_SUBST(SOUNDOBJ)
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@ -273,6 +278,10 @@ case "$host" in
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*-*-hpux*)
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CFLAGS="$CFLAGS -D_XOPEN_SOURCE_EXTENDED"
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;;
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*-*-irix6.5*)
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LIBS="$LIBS -L$ssldir/lib32 -lcrypto"
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CFLAGS="$CFLAGS -D__SGI_IRIX__"
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;;
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esac
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358
rdpsnd_sgi.c
Normal file
358
rdpsnd_sgi.c
Normal file
@ -0,0 +1,358 @@
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/*
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rdesktop: A Remote Desktop Protocol client.
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Sound Channel Process Functions - SGI/IRIX
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Copyright (C) Matthew Chapman 2003
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Copyright (C) GuoJunBo guojunbo@ict.ac.cn 2003
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Copyright (C) Michael Gernoth mike@zerfleddert.de 2003
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This program is free software; you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation; either version 2 of the License, or
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(at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program; if not, write to the Free Software
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Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
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*/
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#include "rdesktop.h"
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#include <errno.h>
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#include <dmedia/audio.h>
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#define IRIX_DEBUG 1
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#define IRIX_MAX_VOL 65535
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#define MAX_QUEUE 10
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int g_dsp_fd;
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ALconfig audioconfig;
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ALport output_port;
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BOOL g_dsp_busy = False;
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static BOOL g_swapaudio;
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static int g_snd_rate;
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static BOOL g_swapaudio;
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static short g_samplewidth;
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static int width = AL_SAMPLE_16;
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double min_volume, max_volume, volume_range;
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int resource, maxFillable;
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int combinedFrameSize;
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static struct audio_packet
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{
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struct stream s;
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uint16 tick;
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uint8 index;
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} packet_queue[MAX_QUEUE];
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static unsigned int queue_hi, queue_lo;
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BOOL
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wave_out_open(void)
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{
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ALparamInfo pinfo;
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#if (defined(IRIX_DEBUG))
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fprintf(stderr, "wave_out_open: begin\n");
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#endif
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if ( alGetParamInfo(AL_DEFAULT_OUTPUT, AL_GAIN, &pinfo) < 0 )
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{
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fprintf(stderr, "wave_out_open: alGetParamInfo failed: %s\n",
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alGetErrorString(oserror()));
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}
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min_volume = alFixedToDouble(pinfo.min.ll);
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max_volume = alFixedToDouble(pinfo.max.ll);
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volume_range = (max_volume - min_volume);
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#if (defined(IRIX_DEBUG))
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fprintf(stderr, "wave_out_open: minvol = %lf, maxvol= %lf, range = %lf.\n",
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min_volume, max_volume, volume_range);
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#endif
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queue_lo = queue_hi = 0;
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audioconfig = alNewConfig();
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if (audioconfig < 0) {
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fprintf(stderr, "wave_out_open: alNewConfig failed: %s\n",
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alGetErrorString(oserror()));
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return False;
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}
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output_port = alOpenPort("rdpsnd", "w", 0);
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if (output_port == (ALport) 0) {
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fprintf(stderr, "wave_out_open: alOpenPort failed: %s\n",
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alGetErrorString(oserror()));
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return False;
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}
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#if (defined(IRIX_DEBUG))
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fprintf(stderr, "wave_out_open: returning\n");
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#endif
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return True;
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}
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void
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wave_out_close(void)
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{
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/* Ack all remaining packets */
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#if (defined(IRIX_DEBUG))
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fprintf(stderr, "wave_out_close: begin\n");
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#endif
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while (queue_lo != queue_hi)
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{
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rdpsnd_send_completion(packet_queue[queue_lo].tick, packet_queue[queue_lo].index);
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free(packet_queue[queue_lo].s.data);
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queue_lo = (queue_lo + 1) % MAX_QUEUE;
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}
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alDiscardFrames(output_port, 0);
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alClosePort(output_port);
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alFreeConfig(audioconfig);
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#if (defined(IRIX_DEBUG))
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fprintf(stderr, "wave_out_close: returning\n");
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#endif
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}
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BOOL
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wave_out_format_supported(WAVEFORMATEX * pwfx)
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{
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if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
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return False;
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if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
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return False;
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if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
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return False;
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return True;
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}
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BOOL
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wave_out_set_format(WAVEFORMATEX * pwfx)
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{
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int channels;
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int frameSize, channelCount;
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ALpv params;
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#if (defined(IRIX_DEBUG))
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fprintf(stderr, "wave_out_set_format: init...\n");
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#endif
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/* limited support to configure an opened audio port in IRIX */
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/* have to reopen the audio port, using same config */
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alClosePort(output_port);
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g_swapaudio = False;
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if (pwfx->wBitsPerSample == 8)
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width = AL_SAMPLE_8;
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else if (pwfx->wBitsPerSample == 16) {
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width = AL_SAMPLE_16;
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/* Do we need to swap the 16bit values? (Are we BigEndian) */
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#if (defined(IRIX_DEBUG))
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g_swapaudio = 1;
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#else
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g_swapaudio = 0;
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#endif
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}
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g_samplewidth = pwfx->wBitsPerSample / 8;
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channels = pwfx->nChannels;
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g_snd_rate = pwfx->nSamplesPerSec;
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alSetSampFmt(audioconfig, AL_SAMPFMT_TWOSCOMP);
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alSetWidth(audioconfig, width);
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alSetChannels(audioconfig, channels);
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output_port = alOpenPort("rdpsnd", "w", audioconfig);
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if (output_port == (ALport) 0) {
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fprintf(stderr, "wave_out_set_format: alOpenPort failed: %s\n",
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alGetErrorString(oserror()));
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return False;
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}
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resource = alGetResource(output_port);
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maxFillable = alGetFillable(output_port);
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channelCount = alGetChannels(audioconfig);
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frameSize = alGetWidth(audioconfig);
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if (frameSize == 0 || channelCount == 0)
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{
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fprintf(stderr, "wave_out_set_format: bad frameSize or channelCount\n");
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return False;
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}
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combinedFrameSize = frameSize * channelCount;
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params.param = AL_RATE;
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params.value.ll = (long long) g_snd_rate << 32;
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if (alSetParams(resource, ¶ms, 1) < 0)
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{
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fprintf(stderr, "wave_set_format: alSetParams failed: %s\n",
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alGetErrorString(oserror()));
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return False;
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}
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if( params.sizeOut < 0 )
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{
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fprintf(stderr, "wave_set_format: invalid rate %d\n", g_snd_rate);
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return False;
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}
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#if (defined(IRIX_DEBUG))
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fprintf(stderr, "wave_out_set_format: returning...\n");
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#endif
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return True;
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}
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void
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wave_out_volume(uint16 left, uint16 right)
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{
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double gainleft, gainright;
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ALpv pv[1];
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ALfixed gain[8];
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#if (defined(IRIX_DEBUG))
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fprintf(stderr, "wave_out_volume: begin\n");
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fprintf(stderr, "left='%d', right='%d'\n", left, right);
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#endif
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gainleft = (double) left / IRIX_MAX_VOL;
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gainright = (double) right / IRIX_MAX_VOL;
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gain[0] = alDoubleToFixed(min_volume + gainleft * volume_range);
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gain[1] = alDoubleToFixed(min_volume + gainright * volume_range);
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pv[0].param = AL_GAIN;
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pv[0].value.ptr = gain;
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pv[0].sizeIn = 8;
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if( alSetParams(AL_DEFAULT_OUTPUT, pv, 1) < 0)
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{
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fprintf(stderr, "wave_out_volume: alSetParams failed: %s\n", alGetErrorString(oserror()));
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return;
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}
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#if (defined(IRIX_DEBUG))
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fprintf(stderr, "wave_out_volume: returning\n");
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#endif
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}
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void
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wave_out_write(STREAM s, uint16 tick, uint8 index)
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{
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struct audio_packet *packet = &packet_queue[queue_hi];
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unsigned int next_hi = (queue_hi + 1) % MAX_QUEUE;
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if (next_hi == queue_lo)
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{
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fprintf(stderr, "No space to queue audio packet\n");
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return;
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}
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queue_hi = next_hi;
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packet->s = *s;
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packet->tick = tick;
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packet->index = index;
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packet->s.p += 4;
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/* we steal the data buffer from s, give it a new one */
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s->data = malloc(s->size);
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if (!g_dsp_busy)
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wave_out_play();
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}
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void
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wave_out_play(void)
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{
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struct audio_packet *packet;
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ssize_t len;
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unsigned int i;
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uint8 swap;
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STREAM out;
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static long startedat_us;
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static long startedat_s;
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static BOOL started = False;
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static BOOL swapped = False;
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struct timeval tv;
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int gf;
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static long long temp;
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while (1)
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{
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if (queue_lo == queue_hi)
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{
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g_dsp_busy = False;
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return;
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}
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packet = &packet_queue[queue_lo];
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out = &packet->s;
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/* Swap the current packet, but only once */
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if (g_swapaudio && !swapped)
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{
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for (i = 0; i < out->end - out->p; i += 2)
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{
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swap = *(out->p + i);
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*(out->p + i) = *(out->p + i + 1);
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*(out->p + i + 1) = swap;
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}
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swapped = True;
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}
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if (!started)
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{
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gettimeofday(&tv, NULL);
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startedat_us = tv.tv_usec;
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startedat_s = tv.tv_sec;
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started = True;
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}
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len = out->end - out->p;
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gf = alGetFillable(output_port);
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if (len > gf) {
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//len = gf * combinedFrameSize;
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#if (defined(IRIX_DEBUG))
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//fprintf(stderr,"Fillable...\n");
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#endif
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}
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alWriteFrames(output_port, out->p, len/combinedFrameSize);
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out->p += len;
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if (out->p == out->end)
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{
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long long duration;
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long elapsed;
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gettimeofday(&tv, NULL);
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duration = (out->size * (1000000 / (g_samplewidth * g_snd_rate)));
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elapsed = (tv.tv_sec - startedat_s) * 1000000
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+ (tv.tv_usec - startedat_us);
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/* 7/10 is not good for IRIX audio port, 4x/100 is suitable */
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if (elapsed >= (duration * 485) / 1000)
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{
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rdpsnd_send_completion(packet->tick, packet->index);
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free(out->data);
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queue_lo = (queue_lo + 1) % MAX_QUEUE;
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started = False;
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swapped = False;
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}
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else
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{
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#if (defined(IRIX_DEBUG))
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//fprintf(stderr,"Busy playing...\n");
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#endif
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g_dsp_busy = True;
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return;
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}
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}
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}
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}
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