📄 rdpsnd_sgi.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, ¶ms, 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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