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/* |
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rdesktop: A Remote Desktop Protocol client. |
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Sound Channel Process Functions - Open Sound System |
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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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|
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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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|
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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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|
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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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|
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#include "rdesktop.h" |
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#include <unistd.h> |
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#include <fcntl.h> |
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#include <errno.h> |
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#include <sys/time.h> |
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#include <sys/ioctl.h> |
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#include <sys/soundcard.h> |
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|
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#define MAX_QUEUE 10 |
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|
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int g_dsp_fd; |
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BOOL g_dsp_busy = False; |
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static int g_snd_rate; |
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static short g_samplewidth; |
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|
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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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|
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BOOL |
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wave_out_open(void) |
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{ |
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char *dsp_dev = getenv("AUDIODEV"); |
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|
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if (dsp_dev == NULL) |
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{ |
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dsp_dev = "/dev/dsp"; |
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} |
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|
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if ((g_dsp_fd = open(dsp_dev, O_WRONLY | O_NONBLOCK)) == -1) |
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{ |
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perror(dsp_dev); |
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return False; |
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} |
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|
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/* Non-blocking so that user interface is responsive */ |
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fcntl(g_dsp_fd, F_SETFL, fcntl(g_dsp_fd, F_GETFL) | O_NONBLOCK); |
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return True; |
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} |
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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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close(g_dsp_fd); |
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} |
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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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|
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return True; |
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} |
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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, format; |
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|
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ioctl(g_dsp_fd, SNDCTL_DSP_RESET, NULL); |
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ioctl(g_dsp_fd, SNDCTL_DSP_SYNC, NULL); |
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|
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if (pwfx->wBitsPerSample == 8) |
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format = AFMT_U8; |
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else if (pwfx->wBitsPerSample == 16) |
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format = AFMT_S16_LE; |
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|
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g_samplewidth = pwfx->wBitsPerSample / 8; |
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|
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if (ioctl(g_dsp_fd, SNDCTL_DSP_SETFMT, &format) == -1) |
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{ |
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perror("SNDCTL_DSP_SETFMT"); |
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close(g_dsp_fd); |
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return False; |
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} |
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|
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channels = pwfx->nChannels; |
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if (ioctl(g_dsp_fd, SNDCTL_DSP_CHANNELS, &channels) == -1) |
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{ |
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perror("SNDCTL_DSP_CHANNELS"); |
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close(g_dsp_fd); |
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return False; |
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} |
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|
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if (channels == 2) |
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{ |
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g_samplewidth *= 2; |
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} |
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|
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g_snd_rate = pwfx->nSamplesPerSec; |
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if (ioctl(g_dsp_fd, SNDCTL_DSP_SPEED, &g_snd_rate) == -1) |
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{ |
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perror("SNDCTL_DSP_SPEED"); |
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close(g_dsp_fd); |
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return False; |
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} |
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|
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return True; |
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} |
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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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static BOOL use_dev_mixer = False; |
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uint32 volume; |
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int fd_mix = -1; |
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|
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volume = left / (65536 / 100); |
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volume |= right / (65536 / 100) << 8; |
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|
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if (use_dev_mixer) |
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{ |
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if ((fd_mix = open("/dev/mixer", O_RDWR | O_NONBLOCK)) == -1) |
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{ |
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perror("open /dev/mixer"); |
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return; |
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} |
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|
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if (ioctl(fd_mix, MIXER_WRITE(SOUND_MIXER_PCM), &volume) == -1) |
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{ |
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perror("MIXER_WRITE(SOUND_MIXER_PCM)"); |
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return; |
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} |
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|
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close(fd_mix); |
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} |
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|
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if (ioctl(g_dsp_fd, MIXER_WRITE(SOUND_MIXER_PCM), &volume) == -1) |
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{ |
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perror("MIXER_WRITE(SOUND_MIXER_PCM)"); |
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use_dev_mixer = True; |
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return; |
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} |
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} |
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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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|
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if (next_hi == queue_lo) |
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{ |
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error("No space to queue audio packet\n"); |
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return; |
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} |
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|
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queue_hi = next_hi; |
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|
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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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|
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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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|
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if (!g_dsp_busy) |
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wave_out_play(); |
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} |
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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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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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struct timeval tv; |
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|
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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 = 0; |
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return; |
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} |
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|
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packet = &packet_queue[queue_lo]; |
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out = &packet->s; |
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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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|
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len = write(g_dsp_fd, out->p, out->end - out->p); |
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if (len == -1) |
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{ |
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if (errno != EWOULDBLOCK) |
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perror("write audio"); |
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g_dsp_busy = 1; |
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return; |
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} |
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|
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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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|
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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 + (tv.tv_usec - startedat_us); |
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|
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if (elapsed >= (duration * 85) / 100) |
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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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} |
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else |
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{ |
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g_dsp_busy = 1; |
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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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