-#if defined(AUDIO_UNIX_NATIVE)
-
-static void CopySampleToMixingBuffer(SoundControl *snd_ctrl,
- int sample_pos, int sample_size,
- short *buffer_base_ptr, int buffer_pos,
- int num_output_channels)
-{
- 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)
- {
- 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 */
- {
- 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[DEFAULT_AUDIO_FRAGMENT_SIZE];
- static int 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)
- return;
-
- if (audio.device_fd < 0)
- {
- if ((audio.device_fd = OpenAudioDevice(audio.device_name)) < 0)
- return;
-
- InitAudioDevice(&afmt);
- }
-
- stereo = afmt.stereo;
- fragment_size = afmt.fragment_size;
- sample_bytes = (afmt.format & AUDIO_FORMAT_U8 ? 1 : 2);
- num_output_channels = (stereo ? 2 : 1);
- max_sample_size = fragment_size / (num_output_channels * sample_bytes);
-
- /* first clear the last premixing buffer */
- clear_mem(premix_last_buffer,
- max_sample_size * num_output_channels * sizeof(int));
-
- for (i = 0; i < audio.num_channels; i++)
- {
- void *sample_ptr;
- int sample_len;
- int sample_pos;
- int sample_size;
-
- if (!mixer[i].active)
- continue;
-
- if (Mixer_ChannelExpired(i))
- {
- Mixer_StopChannel(i);
- continue;
- }
-
- /* pointer, lenght and actual playing position of sound sample */
- sample_ptr = mixer[i].data_ptr;
- sample_len = mixer[i].data_len;
- sample_pos = mixer[i].playing_pos;
- sample_size = MIN(max_sample_size, sample_len - sample_pos);
- mixer[i].playing_pos += sample_size;
-
- /* copy original sample to first mixing buffer */
- CopySampleToMixingBuffer(&mixer[i], sample_pos, sample_size,
- 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)
- {
- while (sample_size < max_sample_size)
- {
- int restarted_sample_size =
- MIN(max_sample_size - sample_size, sample_len);
-
- 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;
- }
- }
-
- /* decrease volume if sound is fading out */
- if (IS_FADING(mixer[i]) &&
- mixer[i].volume >= SOUND_FADING_VOLUME_THRESHOLD)
- mixer[i].volume -= SOUND_FADING_VOLUME_STEP;
-
- /* adjust volume of actual sound sample */
- if (mixer[i].volume != SOUND_MAX_VOLUME)
- for (j = 0; j < sample_size * num_output_channels; j++)
- premix_first_buffer[j] =
- mixer[i].volume * (int)premix_first_buffer[j] / SOUND_MAX_VOLUME;
-
- /* adjust left and right channel volume due to stereo sound position */
- if (stereo)
- {
- int left_volume = SOUND_VOLUME_LEFT(mixer[i].stereo_position);
- int right_volume = SOUND_VOLUME_RIGHT(mixer[i].stereo_position);
-
- for (j = 0; j < sample_size; 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;
- }
- }
-
- /* 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)
- {
- if (IS_LOOP(mixer[i]))
- mixer[i].playing_pos = 0;
- else
- Mixer_StopChannel(i);
- }
- else if (mixer[i].volume <= SOUND_FADING_VOLUME_THRESHOLD)
- Mixer_StopChannel(i);
- }
-
- /* prepare final playing buffer according to system audio format */
- for (i = 0; i < max_sample_size * num_output_channels; i++)
- {
- /* cut off at 17 bit value */
- if (premix_last_buffer[i] < -65535)
- premix_last_buffer[i] = -65535;
- else if (premix_last_buffer[i] > 65535)
- premix_last_buffer[i] = 65535;
-
- /* shift to 16 bit value */
- premix_last_buffer[i] >>= 1;
-
- if (afmt.format & AUDIO_FORMAT_U8)
- {
- playing_buffer[i] = (premix_last_buffer[i] >> 8) ^ 0x80;
- }
- else if (afmt.format & AUDIO_FORMAT_LE) /* 16 bit */
- {
- playing_buffer[2 * i + 0] = premix_last_buffer[i] & 0xff;
- playing_buffer[2 * i + 1] = premix_last_buffer[i] >> 8;
- }
- else /* big endian */
- {
- playing_buffer[2 * i + 0] = premix_last_buffer[i] >> 8;
- playing_buffer[2 * i + 1] = premix_last_buffer[i] & 0xff;
- }
- }
-
- /* finally play the sound fragment */
- write(audio.device_fd, playing_buffer, fragment_size);
-
- if (!mixer_active_channels)
- CloseAudioDevice(&audio.device_fd);
-}
-
-#else /* !AUDIO_STREAMING_DSP */
-
-static int Mixer_Main_SimpleAudio(SoundControl snd_ctrl)
-{
- 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;
- int sample_size;
- int i, j;
-
- i = 1;
-
- /* pointer, lenght and actual playing position of sound sample */
- sample_ptr = mixer[i].data_ptr;
- sample_len = mixer[i].data_len;
- sample_pos = mixer[i].playing_pos;
- sample_size = MIN(max_sample_size, sample_len - sample_pos);
- mixer[i].playing_pos += sample_size;
-
- /* copy original sample to first mixing buffer */
- CopySampleToMixingBuffer(&mixer[i], sample_pos, sample_size,
- premix_first_buffer, 0, num_output_channels);
-
- /* adjust volume of actual sound sample */
- if (mixer[i].volume != SOUND_MAX_VOLUME)
- for (j = 0; j < sample_size; j++)
- premix_first_buffer[j] =
- mixer[i].volume * (int)premix_first_buffer[j] / SOUND_MAX_VOLUME;
-
- /* might be needed for u-law /dev/audio */
- for (j = 0; j < sample_size; j++)
- playing_buffer[j] =
- linear_to_ulaw(premix_first_buffer[j]);
-
- /* delete completed sound entries from the mixer */
- if (mixer[i].playing_pos >= mixer[i].data_len)
- Mixer_StopChannel(i);
-
- for (i = 0; i < sample_size; i++)
- playing_buffer[i] = (premix_first_buffer[i] >> 8) ^ 0x80;
-
- /* finally play the sound fragment */
- write(audio.device_fd, playing_buffer, sample_size);
-
- return sample_size;
-}
-#endif /* !AUDIO_STREAMING_DSP */
-
-void Mixer_Main()
-{
- SoundControl snd_ctrl;
- fd_set mixer_fdset;
-
- close(audio.mixer_pipe[1]); /* no writing into pipe needed */
-
- Mixer_InitChannels();
-
-#if defined(PLATFORM_HPUX)
- InitAudioDevice(&afmt);
-#endif
-
- FD_ZERO(&mixer_fdset);
- FD_SET(audio.mixer_pipe[0], &mixer_fdset);
-
- while (1) /* wait for sound playing commands from client */
- {
- struct timeval delay = { 0, 0 };
-
- FD_SET(audio.mixer_pipe[0], &mixer_fdset);
- select(audio.mixer_pipe[0] + 1, &mixer_fdset, NULL, NULL, NULL);
- if (!FD_ISSET(audio.mixer_pipe[0], &mixer_fdset))
- continue;
-
- ReadSoundControlFromMainProcess(&snd_ctrl);
-
- HandleSoundRequest(snd_ctrl);
-
-#if defined(AUDIO_STREAMING_DSP)
-
- while (mixer_active_channels &&
- select(audio.mixer_pipe[0] + 1,
- &mixer_fdset, NULL, NULL, &delay) < 1)
- {
- FD_SET(audio.mixer_pipe[0], &mixer_fdset);
-
- Mixer_Main_DSP();
- }
-
-#else /* !AUDIO_STREAMING_DSP */
-
- if (!snd_ctrl.active || IS_LOOP(snd_ctrl) ||
- (audio.device_fd = OpenAudioDevice(audio.device_name)) < 0)
- continue;
-
- InitAudioDevice(&afmt);
-
- delay.tv_sec = 0;
- delay.tv_usec = 0;
-
- while (mixer_active_channels &&
- select(audio.mixer_pipe[0] + 1,
- &mixer_fdset, NULL, NULL, &delay) < 1)
- {
- int wait_percent = 90; /* wait 90% of the real playing time */
- int sample_size;
-
- FD_SET(audio.mixer_pipe[0], &mixer_fdset);
-
- sample_size = Mixer_Main_SimpleAudio(snd_ctrl);
-
- delay.tv_sec = 0;
- delay.tv_usec =
- ((sample_size * 10 * wait_percent) / afmt.sample_rate) * 1000;
- }
-
- CloseAudioDevice(&audio.device_fd);
-
- Mixer_InitChannels(); /* remove all sounds from mixer */
-
-#endif /* !AUDIO_STREAMING_DSP */
- }
-}
-#endif /* AUDIO_UNIX_NATIVE */
-
-
-#if defined(AUDIO_UNIX_NATIVE) && !defined(AUDIO_STREAMING_DSP)
-
-/* these two are stolen from "sox"... :) */
-
-/*
-** This routine converts from linear to ulaw.
-**
-** Craig Reese: IDA/Supercomputing Research Center
-** Joe Campbell: Department of Defense
-** 29 September 1989
-**
-** References:
-** 1) CCITT Recommendation G.711 (very difficult to follow)
-** 2) "A New Digital Technique for Implementation of Any
-** Continuous PCM Companding Law," Villeret, Michel,
-** et al. 1973 IEEE Int. Conf. on Communications, Vol 1,
-** 1973, pg. 11.12-11.17
-** 3) MIL-STD-188-113,"Interoperability and Performance Standards
-** for Analog-to_Digital Conversion Techniques,"
-** 17 February 1987
-**
-** Input: Signed 16 bit linear sample
-** Output: 8 bit ulaw sample
-*/
-
-#define ZEROTRAP /* turn on the trap as per the MIL-STD */
-#define BIAS 0x84 /* define the add-in bias for 16 bit samples */
-#define CLIP 32635
-
-static unsigned char linear_to_ulaw(int sample)
-{
- static int exp_lut[256] =
- {
- 0,0,1,1,2,2,2,2,3,3,3,3,3,3,3,3,
- 4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,
- 5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,
- 5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,
- 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
- 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
- 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
- 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
- 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
- 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
- 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
- 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
- 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
- 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
- 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
- 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7
- };
-
- int sign, exponent, mantissa;
- unsigned char ulawbyte;
-
- /* Get the sample into sign-magnitude. */
- sign = (sample >> 8) & 0x80; /* set aside the sign */
- if (sign != 0)
- sample = -sample; /* get magnitude */
- if (sample > CLIP)
- sample = CLIP; /* clip the magnitude */
-
- /* Convert from 16 bit linear to ulaw. */
- sample = sample + BIAS;
- exponent = exp_lut[( sample >> 7 ) & 0xFF];
- mantissa = ( sample >> ( exponent + 3 ) ) & 0x0F;
- ulawbyte = ~ ( sign | ( exponent << 4 ) | mantissa );
-#ifdef ZEROTRAP
- if (ulawbyte == 0)
- ulawbyte = 0x02; /* optional CCITT trap */
-#endif
-
- return(ulawbyte);
-}
-
-/*
-** This routine converts from ulaw to 16 bit linear.
-**
-** Craig Reese: IDA/Supercomputing Research Center
-** 29 September 1989
-**
-** References:
-** 1) CCITT Recommendation G.711 (very difficult to follow)
-** 2) MIL-STD-188-113,"Interoperability and Performance Standards
-** for Analog-to_Digital Conversion Techniques,"
-** 17 February 1987
-**
-** Input: 8 bit ulaw sample
-** Output: signed 16 bit linear sample
-*/
-
-static int ulaw_to_linear(unsigned char ulawbyte)
-{
- static int exp_lut[8] = { 0, 132, 396, 924, 1980, 4092, 8316, 16764 };
- int sign, exponent, mantissa, sample;
-
- ulawbyte = ~ ulawbyte;
- sign = ( ulawbyte & 0x80 );
- exponent = ( ulawbyte >> 4 ) & 0x07;
- mantissa = ulawbyte & 0x0F;
- sample = exp_lut[exponent] + ( mantissa << ( exponent + 3 ) );
- if (sign != 0)
- sample = -sample;
-
- return(sample);
-}
-#endif /* AUDIO_UNIX_NATIVE && !AUDIO_STREAMING_DSP */
-