static int stereo_volume[PSND_MAX_LEFT2RIGHT+1];
static char premix_first_buffer[SND_BLOCKSIZE];
#if defined(AUDIO_STREAMING_DSP)
-static char premix_left_buffer[SND_BLOCKSIZE];
-static char premix_right_buffer[SND_BLOCKSIZE];
-static int premix_last_buffer[SND_BLOCKSIZE];
+static short premix_second_buffer[SND_BLOCKSIZE];
+static short premix_left_buffer[SND_BLOCKSIZE];
+static short premix_right_buffer[SND_BLOCKSIZE];
+static long premix_last_buffer[SND_BLOCKSIZE];
#endif
-static unsigned char playing_buffer[SND_BLOCKSIZE];
+static short playing_buffer[SND_BLOCKSIZE];
#endif
/* forward declaration of internal functions */
struct timeval delay = { 0, 0 };
byte *sample_ptr;
long sample_size;
- static long max_sample_size = 0;
- static long fragment_size = DEFAULT_AUDIO_FRAGMENT_SIZE;
+ static int max_sample_size = 0;
+ static int fragment_size = DEFAULT_AUDIO_FRAGMENT_SIZE;
int sample_rate = DEFAULT_AUDIO_SAMPLE_RATE;
static boolean stereo = TRUE;
#elif defined(PLATFORM_NETBSD)
stereo = InitAudioDevice_NetBSD(fragment_size, sample_rate);
#endif
- max_sample_size = fragment_size / (stereo ? 2 : 1);
+ max_sample_size = fragment_size / ((stereo ? 2 : 1) * sizeof(short));
}
if (snd_ctrl.active) /* new sound has arrived */
FD_SET(audio.soundserver_pipe[0], &sound_fdset);
/* first clear the last premixing buffer */
- memset(premix_last_buffer, 0, fragment_size * sizeof(int));
+ memset(premix_last_buffer, 0, fragment_size * sizeof(short));
- for(i=0;i<MAX_SOUNDS_PLAYING;i++)
+ for(i=0; i<MAX_SOUNDS_PLAYING; i++)
{
int j;
sample_size = max_sample_size;
}
+#if 0
+ /* expand sample from 8 to 16 bit */
+ for(j=0; j<sample_size; j++)
+ premix_second_buffer[j] = premix_first_buffer[j] << 8;
+#endif
+
/* decrease volume if sound is fading out */
if (playlist[i].fade_sound &&
playlist[i].volume >= SOUND_FADING_VOLUME_THRESHOLD)
if (stereo)
{
int middle_pos = PSND_MAX_LEFT2RIGHT/2;
- int left_volume = stereo_volume[middle_pos +playlist[i].stereo];
- int right_volume = stereo_volume[middle_pos -playlist[i].stereo];
+ int left_volume = stereo_volume[middle_pos + playlist[i].stereo];
+ int right_volume= stereo_volume[middle_pos - playlist[i].stereo];
for(j=0; j<sample_size; j++)
{
premix_right_buffer[j] =
(right_volume * (int)premix_first_buffer[j])
>> PSND_MAX_LEFT2RIGHT_BITS;
- premix_last_buffer[2*j+0] += premix_left_buffer[j];
- premix_last_buffer[2*j+1] += premix_right_buffer[j];
+
+ premix_last_buffer[2 * j + 0] += premix_left_buffer[j];
+ premix_last_buffer[2 * j + 1] += premix_right_buffer[j];
}
}
else
{
- for(j=0;j<sample_size;j++)
+ for(j=0; j<sample_size; j++)
premix_last_buffer[j] += (int)premix_first_buffer[j];
}
}
/* put last mixing buffer to final playing buffer */
+#if 0
for(i=0; i<fragment_size; i++)
{
- if (premix_last_buffer[i]<-255)
+ if (premix_last_buffer[i] < -255)
playing_buffer[i] = 0;
- else if (premix_last_buffer[i]>255)
+ else if (premix_last_buffer[i] > 255)
playing_buffer[i] = 255;
else
- playing_buffer[i] = (premix_last_buffer[i]>>1)^0x80;
+ playing_buffer[i] = (premix_last_buffer[i] >> 1) ^ 0x80;
}
+#else
+ for(i=0; i<fragment_size; i++)
+ {
+ if (premix_last_buffer[i] < -255)
+ playing_buffer[i] = -127;
+ else if (premix_last_buffer[i] > 255)
+ playing_buffer[i] = 127;
+ else
+ playing_buffer[i] = (premix_last_buffer[i] >> 1);
+
+ playing_buffer[i] <<= 8;
+ }
+#endif
/* finally play the sound fragment */
write(audio.device_fd, playing_buffer, fragment_size);
/* ------------------------------------------------------------------------- */
#if defined(AUDIO_LINUX_IOCTL)
-static boolean InitAudioDevice_Linux(long fragment_size, int sample_rate)
+static boolean InitAudioDevice_Linux(int fragment_size, int sample_rate)
{
/* "ioctl()" expects pointer to 'int' value for stereo flag
(boolean is defined as 'char', which will not work here) */
+ unsigned int fragment_spec = 0;
+ unsigned int audio_format = 0;
+ int fragment_size_query;
int stereo = TRUE;
- unsigned long fragment_spec = 0;
/* determine logarithm (log2) of the fragment size */
- for (fragment_spec=0; (1 << fragment_spec) < fragment_size;
- fragment_spec++);
+ for (fragment_spec=0; (1 << fragment_spec) < fragment_size; fragment_spec++);
/* use two fragments (play one fragment, prepare the other);
one fragment would result in interrupted audio output, more
Error(ERR_EXIT_SOUND_SERVER,
"cannot set fragment size of /dev/dsp -- no sounds");
+ audio_format = AFMT_S16_LE;
+ if (ioctl(audio.device_fd, SNDCTL_DSP_SETFMT, &audio_format) < 0)
+ Error(ERR_EXIT_SOUND_SERVER,
+ "cannot set audio format of /dev/dsp -- no sounds");
+
/* try if we can use stereo sound */
if (ioctl(audio.device_fd, SNDCTL_DSP_STEREO, &stereo) < 0)
{
"cannot set sample rate of /dev/dsp -- no sounds");
/* get the real fragmentation size; this should return 512 */
- if (ioctl(audio.device_fd, SNDCTL_DSP_GETBLKSIZE, &fragment_size) < 0)
+ if (ioctl(audio.device_fd, SNDCTL_DSP_GETBLKSIZE, &fragment_size_query) < 0)
Error(ERR_EXIT_SOUND_SERVER,
"cannot get fragment size of /dev/dsp -- no sounds");
+ if (fragment_size_query != fragment_size)
+ Error(ERR_EXIT_SOUND_SERVER,
+ "cannot set fragment size of /dev/dsp -- no sounds");
return (boolean)stereo;
}
#endif /* AUDIO_LINUX_IOCTL */
#if defined(PLATFORM_NETBSD)
-static boolean InitAudioDevice_NetBSD(long fragment_size, int sample_rate)
+static boolean InitAudioDevice_NetBSD(int fragment_size, int sample_rate)
{
audio_info_t a_info;
boolean stereo = TRUE;
}
for (i=0; i<snd_info->data_len; i++)
- ((byte *)snd_info->data_ptr)[i] = ((byte *)snd_info->data_ptr)[i] ^ 0x80;
+ ((byte *)snd_info->data_ptr)[i] ^= 0x80;
#endif /* PLATFORM_UNIX */