#define SOUND_FADING_VOLUME_THRESHOLD (SOUND_FADING_VOLUME_STEP * 2)
#endif
-#if !defined(PLATFORM_HPUX)
-#define SND_BLOCKSIZE 4096
-#else
-#define SND_BLOCKSIZE 32768
-#endif
-
#define SND_TYPE_NONE 0
#define SND_TYPE_WAV 1
int format;
void *data_ptr; /* pointer to first sample (8 or 16 bit) */
long data_len; /* number of samples, NOT number of bytes */
+ int num_channels; /* mono: 1 channel, stereo: 2 channels */
};
typedef struct SampleInfo SoundInfo;
typedef struct SampleInfo MusicInfo;
int format;
void *data_ptr; /* pointer to first sample (8 or 16 bit) */
long data_len; /* number of samples, NOT number of bytes */
+ int num_channels; /* mono: 1 channel, stereo: 2 channels */
#if defined(TARGET_ALLEGRO)
int voice;
return;
/* copy sound sample and format information */
- snd_ctrl.type = snd_info->type;
- snd_ctrl.format = snd_info->format;
- snd_ctrl.data_ptr = snd_info->data_ptr;
- snd_ctrl.data_len = snd_info->data_len;
+ snd_ctrl.type = snd_info->type;
+ snd_ctrl.format = snd_info->format;
+ snd_ctrl.data_ptr = snd_info->data_ptr;
+ snd_ctrl.data_len = snd_info->data_len;
+ snd_ctrl.num_channels = snd_info->num_channels;
/* play music samples on a dedicated music channel */
if (IS_MUSIC(snd_ctrl))
static void CopySampleToMixingBuffer(SoundControl *snd_ctrl,
int sample_pos, int sample_size,
- short *buffer_ptr)
+ short *buffer_base_ptr, int buffer_pos,
+ int num_output_channels)
{
- void *sample_ptr = snd_ctrl->data_ptr;
- int i;
+ short *buffer_ptr = buffer_base_ptr + num_output_channels * buffer_pos;
+ int num_channels = snd_ctrl->num_channels;
+ int stepsize = num_channels;
+ int output_stepsize = num_output_channels;
+ int i, j;
if (snd_ctrl->format == AUDIO_FORMAT_U8)
- for (i=0; i<sample_size; i++)
- *buffer_ptr++ =
- ((short)(((byte *)sample_ptr)[sample_pos + i] ^ 0x80)) << 8;
+ {
+ byte *sample_ptr = (byte *)snd_ctrl->data_ptr + num_channels * sample_pos;
+
+ for (i=0; i<num_output_channels; i++)
+ {
+ int offset = (snd_ctrl->num_channels == 1 ? 0 : i);
+
+ for (j=0; j<sample_size; j++)
+ buffer_ptr[output_stepsize * j + i] =
+ ((short)(sample_ptr[stepsize * j + offset] ^ 0x80)) << 8;
+ }
+ }
else /* AUDIO_FORMAT_S16 */
- for (i=0; i<sample_size; i++)
- *buffer_ptr++ =
- ((short *)sample_ptr)[sample_pos + i];
+ {
+ short *sample_ptr= (short *)snd_ctrl->data_ptr + num_channels * sample_pos;
+
+ for (i=0; i<num_output_channels; i++)
+ {
+ int offset = (snd_ctrl->num_channels == 1 ? 0 : i);
+
+ for (j=0; j<sample_size; j++)
+ buffer_ptr[output_stepsize * j + i] =
+ sample_ptr[stepsize * j + offset];
+ }
+ }
}
#if defined(AUDIO_STREAMING_DSP)
static void Mixer_Main_DSP()
{
- static short premix_first_buffer[SND_BLOCKSIZE];
- static short premix_left_buffer[SND_BLOCKSIZE];
- static short premix_right_buffer[SND_BLOCKSIZE];
- static long premix_last_buffer[SND_BLOCKSIZE];
- static byte playing_buffer[SND_BLOCKSIZE];
+ static short premix_first_buffer[DEFAULT_AUDIO_FRAGMENT_SIZE];
+ static long premix_last_buffer[DEFAULT_AUDIO_FRAGMENT_SIZE];
+ static byte playing_buffer[DEFAULT_AUDIO_FRAGMENT_SIZE];
boolean stereo;
int fragment_size;
int sample_bytes;
int max_sample_size;
+ int num_output_channels;
int i, j;
if (!mixer_active_channels)
stereo = afmt.stereo;
fragment_size = afmt.fragment_size;
sample_bytes = (afmt.format & AUDIO_FORMAT_U8 ? 1 : 2);
- max_sample_size = fragment_size / ((stereo ? 2 : 1) * sample_bytes);
+ num_output_channels = (stereo ? 2 : 1);
+ max_sample_size = fragment_size / (num_output_channels * sample_bytes);
/* first clear the last premixing buffer */
memset(premix_last_buffer, 0,
- max_sample_size * (stereo ? 2 : 1) * sizeof(long));
+ max_sample_size * num_output_channels * sizeof(long));
for(i=0; i<audio.num_channels; i++)
{
/* copy original sample to first mixing buffer */
CopySampleToMixingBuffer(&mixer[i], sample_pos, sample_size,
- premix_first_buffer);
+ premix_first_buffer, 0, num_output_channels);
/* are we about to restart a looping sound? */
if (IS_LOOP(mixer[i]) && sample_size < max_sample_size)
int restarted_sample_size =
MIN(max_sample_size - sample_size, sample_len);
- if (mixer[i].format == AUDIO_FORMAT_U8)
- for (j=0; j<restarted_sample_size; j++)
- premix_first_buffer[sample_size + j] =
- ((short)(((byte *)sample_ptr)[j] ^ 0x80)) << 8;
- else
- for (j=0; j<restarted_sample_size; j++)
- premix_first_buffer[sample_size + j] =
- ((short *)sample_ptr)[j];
+ CopySampleToMixingBuffer(&mixer[i], 0, restarted_sample_size,
+ premix_first_buffer, sample_size,
+ num_output_channels);
mixer[i].playing_pos = restarted_sample_size;
sample_size += restarted_sample_size;
/* adjust volume of actual sound sample */
if (mixer[i].volume != SOUND_MAX_VOLUME)
- for(j=0; j<sample_size; j++)
+ for(j=0; j<sample_size * num_output_channels; j++)
premix_first_buffer[j] =
mixer[i].volume * (long)premix_first_buffer[j] / SOUND_MAX_VOLUME;
- /* fill the last mixing buffer with stereo or mono sound */
+ /* adjust left and right channel volume due to stereo sound position */
if (stereo)
{
int left_volume = SOUND_VOLUME_LEFT(mixer[i].stereo_position);
for(j=0; j<sample_size; j++)
{
- premix_left_buffer[j] =
- left_volume * premix_first_buffer[j] / SOUND_MAX_LEFT2RIGHT;
- premix_right_buffer[j] =
- right_volume * premix_first_buffer[j] / SOUND_MAX_LEFT2RIGHT;
-
- premix_last_buffer[2 * j + 0] += premix_left_buffer[j];
- premix_last_buffer[2 * j + 1] += premix_right_buffer[j];
+ premix_first_buffer[2 * j + 0] =
+ left_volume * premix_first_buffer[2 * j + 0] / SOUND_MAX_LEFT2RIGHT;
+ premix_first_buffer[2 * j + 1] =
+ right_volume * premix_first_buffer[2 * j + 1] / SOUND_MAX_LEFT2RIGHT;
}
}
- else
- {
- for(j=0; j<sample_size; j++)
- premix_last_buffer[j] += premix_first_buffer[j];
- }
+
+ /* fill the last mixing buffer with stereo or mono sound */
+ for(j=0; j<sample_size * num_output_channels; j++)
+ premix_last_buffer[j] += premix_first_buffer[j];
/* delete completed sound entries from the mixer */
if (mixer[i].playing_pos >= mixer[i].data_len)
}
/* prepare final playing buffer according to system audio format */
- for(i=0; i<max_sample_size * (stereo ? 2 : 1); i++)
+ for(i=0; i<max_sample_size * num_output_channels; i++)
{
/* cut off at 17 bit value */
if (premix_last_buffer[i] < -65535)
static int Mixer_Main_SimpleAudio(SoundControl snd_ctrl)
{
- static short premix_first_buffer[SND_BLOCKSIZE];
- static byte playing_buffer[SND_BLOCKSIZE];
- int max_sample_size = SND_BLOCKSIZE;
+ static short premix_first_buffer[DEFAULT_AUDIO_FRAGMENT_SIZE];
+ static byte playing_buffer[DEFAULT_AUDIO_FRAGMENT_SIZE];
+ int max_sample_size = DEFAULT_AUDIO_FRAGMENT_SIZE;
+ int num_output_channels = 1;
void *sample_ptr;
int sample_len;
int sample_pos;
/* copy original sample to first mixing buffer */
CopySampleToMixingBuffer(&mixer[i], sample_pos, sample_size,
- premix_first_buffer);
+ premix_first_buffer, 0, num_output_channels);
/* adjust volume of actual sound sample */
if (mixer[i].volume != SOUND_MAX_VOLUME)
#endif
char chunk_name[CHUNK_ID_LEN + 1];
int chunk_size;
+ int data_byte_len;
FILE *file;
#endif
return NULL;
}
- if (header.bits_per_sample != 8 && header.bits_per_sample != 16)
+ if (header.bits_per_sample != 8 &&
+ header.bits_per_sample != 16)
{
Error(ERR_WARN, "sound file '%s': %d bits per sample not supported",
filename, header.bits_per_sample);
}
else if (strcmp(chunk_name, "data") == 0)
{
- snd_info->data_len = chunk_size;
+ data_byte_len = chunk_size;
+
+ snd_info->data_len = data_byte_len;
snd_info->data_ptr = checked_malloc(snd_info->data_len);
/* read sound data */
return NULL;
}
- /* check for odd number of sample bytes (data chunk is word aligned) */
- if ((chunk_size % 2) == 1)
+ /* check for odd number of data bytes (data chunk is word aligned) */
+ if ((data_byte_len % 2) == 1)
ReadUnusedBytesFromFile(file, 1);
}
else /* unknown chunk -- ignore */
snd_info->data_len /= 2; /* correct number of samples */
}
+ snd_info->num_channels = header.num_channels;
+ if (header.num_channels == 2)
+ snd_info->data_len /= 2; /* correct number of samples */
+
+#if 0
+ if (header.num_channels == 1) /* convert mono sound to stereo */
+ {
+ void *buffer_ptr = checked_malloc(data_byte_len * 2);
+ void *sample_ptr = snd_info->data_ptr;
+ int sample_size = snd_info->data_len;
+ int i;
+
+ if (snd_ctrl->format == AUDIO_FORMAT_U8)
+ for (i=0; i<sample_size; i++)
+ *buffer_ptr++ =
+ ((short)(((byte *)sample_ptr)[i] ^ 0x80)) << 8;
+ else /* AUDIO_FORMAT_S16 */
+ for (i=0; i<sample_size; i++)
+ *buffer_ptr++ =
+ ((short *)sample_ptr)[i];
+ }
+#endif
+
#endif /* AUDIO_UNIX_NATIVE */
snd_info->type = SND_TYPE_WAV;