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📄 rdpsnd_sgi.c

📁 rdesktop is a client for Microsoft Windows NT Terminal Server, Windows 2000 Terminal Services, Wind
💻 C
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/* -*- 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., 675 Mass Ave, Cambridge, MA 02139, USA.*/#include "rdesktop.h"#include <errno.h>#include <dmedia/audio.h>/* #define IRIX_DEBUG 1 */#define IRIX_MAX_VOL     65535#define MAX_QUEUE	10int g_dsp_fd;ALconfig audioconfig;ALport output_port;BOOL g_dsp_busy = 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;BOOLwave_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;}voidwave_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}BOOLwave_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;}BOOLwave_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;}voidwave_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}voidwave_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 (!g_dsp_busy)		wave_out_play();}voidwave_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)		{			g_dsp_busy = 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				g_dsp_busy = True;				usleep(10);				return;			}		}	}}

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