Subversion Repositories Games.Chess Giants

// audio.cpp

#include "common.h"

// OpenAL includes

#include "openal/al.h"

#include "openal/alc.h"

// attenuation factor (lower means sounds fade LESS with distance, higher means sounds fade MORE with distance)

#define ATTENUATION_FACTOR 0.02f

// structures used in this module only

typedef struct openal_buffer_s

{

   wchar_t *pathname; // sound pathname (mallocated, if NULL then slot is empty)

   ALshort *openal_samples; // OpenAL samples (16-bit mono)

   ALuint openal_buffer; // associated OpenAL buffer ID

   struct openal_buffer_s *next; // pointer to the next element in list

} openal_buffer_t;

typedef struct openal_source_s

{

   bool is_used; // set to TRUE if this source is used

   ALuint openal_source; // OpenAL sound source

} openal_source_t;

// global variables used in this module only

static ALCdevice *openal_device;

static ALCcontext *openal_context;

static openal_buffer_t *soundbuffers; // mallocated linked list

static openal_source_t *sources; // mallocated

static int source_count;

bool Audio_Init (void)

{

   // this function initializes the audio subsystem (OpenAL)

   openal_device = alcOpenDevice (NULL); // open audio device

   if (openal_device == NULL)

      return (false);

   openal_context = alcCreateContext (openal_device, NULL); // create audio context

   if (openal_context == NULL)

      return (false);

   if (!alcMakeContextCurrent (openal_context)) // select this audio context

      return (false);

   soundbuffers = NULL; // we know no soundbuffer yet

   sources = NULL; // we have no playing source yet

   source_count = 0;

   return (true); // audio subsystem successfully initialized

}

void Audio_Shutdown (void)

{

   // this function shuts down the audio subsystem

   openal_buffer_t *soundbuffer;

   openal_buffer_t *oldbuffer;

   int array_index;

   ALint status;

   // cycle through all sound sources, stop them if needed and delete them

   for (array_index = 0; array_index < source_count; array_index++)

   {

      if (!sources[array_index].is_used)

         continue; // skip unused sources

      alGetSourcei (sources[array_index].openal_source, AL_SOURCE_STATE, &status); // get this source's playing state

      if (status == AL_PLAYING)

         alSourceStop (sources[array_index].openal_source); // stop all playing sources

      alSourcei (sources[array_index].openal_source, AL_BUFFER, NULL); // untie the buffer from this source

      alDeleteSources (1, &sources[array_index].openal_source); // and tell OpenAL to dispose of it

      sources[array_index].is_used = false; // mark this source as unused now

   }

   // cycle through all known sound buffers and delete them

   soundbuffer = soundbuffers;

   while (soundbuffer != NULL)

   {

      alDeleteBuffers (1, &soundbuffer->openal_buffer); // tell OpenAL to dispose of this buffer

      SAFE_free ((void **) &soundbuffer->openal_samples); // free the buffer by our side

      SAFE_free ((void **) &soundbuffer->pathname); // remember this buffer is now empty

      oldbuffer = soundbuffer; // save a pointer to the element we just processed

      soundbuffer = soundbuffer->next; // proceed to the next element in the linked list

      SAFE_free ((void **) &oldbuffer); // and free the element we just processed

   }

   soundbuffers = NULL; // at this point, all elements in the linked list are free

   alcMakeContextCurrent (NULL); // unselect the audio context

   alcDestroyContext (openal_context); // destroy it (only after it's been unselected!)

   alcCloseDevice (openal_device); // and close the audio device

   return; // finished, audio subsystem has been shutdown

}

void Audio_Think (void)

{

   // this function disposes of sound buffers and sources that have finished playing

   int source_index;

   ALint status;

   float angle;

   float sin_pitch;

   float sin_yaw;

   float cos_pitch;

   float cos_yaw;

   ALfloat camera_position[3];

   ALfloat forward_and_up[6];

   // compute the sine and cosine of the pitch component

   angle = current_pitch * TO_RADIANS;

   sin_pitch = sinf (angle);

   cos_pitch = cosf (angle);

   // compute the sine and cosine of the yaw component

   angle = current_yaw * TO_RADIANS;

   sin_yaw = sinf (angle);

   cos_yaw = cosf (angle);

   // build the camera position

   camera_position[0] = (ALfloat) -(cos_pitch * cos_yaw) * current_distance * ATTENUATION_FACTOR;

   camera_position[1] = (ALfloat) -(cos_pitch * sin_yaw) * current_distance * ATTENUATION_FACTOR;

   camera_position[2] = (ALfloat) sin_pitch * current_distance * ATTENUATION_FACTOR;

   // build the camera orientation

   forward_and_up[0] = -camera_position[0]; // forward direction is the opposite of camera position, since the camera is looking at the center of the scene

   forward_and_up[1] = -camera_position[1];

   forward_and_up[2] = -camera_position[2];

   forward_and_up[3] = 0.0f;

   forward_and_up[3] = 0.0f;

   forward_and_up[3] = 1.0f; // FIXME: upwards direction is not quite exact. It depends on the lookdown angle.

   // update the listener's position and orientation

   alListener3f (AL_POSITION, camera_position[0], camera_position[1], camera_position[2]);

   alListener3f (AL_VELOCITY, 0, 0, 0); // TODO: compute velocity dynamically with previous position

   alListenerfv (AL_ORIENTATION, forward_and_up);

   // cycle through all used sources and see if one is no longer playing

   for (source_index = 0; source_index < source_count; source_index++)

   {

      if (!sources[source_index].is_used)

         continue; // skip unused slots

      alGetSourcei (sources[source_index].openal_source, AL_SOURCE_STATE, &status); // get this source's playing state

      if (status == AL_PLAYING)

         continue; // skip sources that are still playing

      alSourcei (sources[source_index].openal_source, AL_BUFFER, NULL); // untie the buffer from this source

      alDeleteSources (1, &sources[source_index].openal_source); // and tell OpenAL to dispose of it

      sources[source_index].is_used = false; // mark this source as unused now

   }

   return; // finished, audio has been handled

}

void Audio_PlaySound (int sound_type, float pos_x, float pos_y, float pos_z)

{

   // helper function to play a sound

   static wchar_t soundfile_path[MAX_PATH];

   openal_buffer_t *soundbuffer;

   buffer_t soundfile;

   unsigned long current_pos;

   uint16_t temp16;

   uint32_t temp32;

   uint32_t blksiz;

   uint32_t chunk_id;

   int sample_count;

   int sample_index;

   int sample_size;

   int channel_count;

   int channel_index;

   int channel_size;

   int sample_value;

   int sample_rate;

   int source_index;

   float pitch;

   if (!options.want_sounds)

      return; // if we want no sound, don't play anything

   // given the type of sound we want, enqueue the right one

   pitch = 1.0f; // assume fixed pitch until told otherwise

   if      (sound_type == SOUNDTYPE_CLICK)       swprintf_s (soundfile_path, WCHAR_SIZEOF (soundfile_path), L"%s/themes/%s/sounds/click.wav", app_path, theme->name);

   else if (sound_type == SOUNDTYPE_ILLEGALMOVE) swprintf_s (soundfile_path, WCHAR_SIZEOF (soundfile_path), L"%s/themes/%s/sounds/illegal.wav", app_path, theme->name);

   else if (sound_type == SOUNDTYPE_VICTORY)     swprintf_s (soundfile_path, WCHAR_SIZEOF (soundfile_path), L"%s/themes/%s/sounds/win.wav", app_path, theme->name);

   else if (sound_type == SOUNDTYPE_DEFEAT)      swprintf_s (soundfile_path, WCHAR_SIZEOF (soundfile_path), L"%s/themes/%s/sounds/lose.wav", app_path, theme->name);

   else if (sound_type == SOUNDTYPE_CHECK)       swprintf_s (soundfile_path, WCHAR_SIZEOF (soundfile_path), L"%s/themes/%s/sounds/check.wav", app_path, theme->name);

   else if (sound_type == SOUNDTYPE_PIECETAKEN)  swprintf_s (soundfile_path, WCHAR_SIZEOF (soundfile_path), L"%s/themes/%s/sounds/take.wav", app_path, theme->name);

   else if (sound_type == SOUNDTYPE_MOVE)

   {

      temp32 = rand () % 6; // there are several movement sounds, pick one at random

      if      (temp32 == 0) swprintf_s (soundfile_path, WCHAR_SIZEOF (soundfile_path), L"%s/themes/%s/sounds/move1.wav", app_path, theme->name);

      else if (temp32 == 1) swprintf_s (soundfile_path, WCHAR_SIZEOF (soundfile_path), L"%s/themes/%s/sounds/move2.wav", app_path, theme->name);

      else if (temp32 == 2) swprintf_s (soundfile_path, WCHAR_SIZEOF (soundfile_path), L"%s/themes/%s/sounds/move3.wav", app_path, theme->name);

      else if (temp32 == 3) swprintf_s (soundfile_path, WCHAR_SIZEOF (soundfile_path), L"%s/themes/%s/sounds/move4.wav", app_path, theme->name);

      else if (temp32 == 4) swprintf_s (soundfile_path, WCHAR_SIZEOF (soundfile_path), L"%s/themes/%s/sounds/move5.wav", app_path, theme->name);

      else                  swprintf_s (soundfile_path, WCHAR_SIZEOF (soundfile_path), L"%s/themes/%s/sounds/move6.wav", app_path, theme->name);

      pitch = 1.0f + ((((float) rand ()) / RAND_MAX) - 0.5f) / 2.0f; // set a random pitch for these sounds between 0.75 and 1.25

   }

   else if (sound_type == SOUNDTYPE_SLIDE)

   {

      swprintf_s(soundfile_path, WCHAR_SIZEOF(soundfile_path), L"%s/themes/%s/sounds/slide.wav", app_path, theme->name);

      pitch = 1.0f + ((((float)rand()) / RAND_MAX) - 0.5f) / 2.0f; // set a random pitch for these sounds between 0.75 and 1.25

   }

   // now cycle through our known OpenAL buffers and see if we already know this one

   for (soundbuffer = soundbuffers; soundbuffer != NULL; soundbuffer = soundbuffer->next)

      if ((soundbuffer->pathname != NULL) && (wcscmp (soundbuffer->pathname, soundfile_path) == 0))

         break; // break as soon as we find it

   // have we NOT found it ? if so, we must create it

   if (soundbuffer == NULL)

   {

      // load the sound file

      Buffer_Initialize (&soundfile);

      if (!Buffer_ReadFromFileW (&soundfile, soundfile_path))

         return; // if unable to load this sound file, give up (FIXME: log something ?)

      // parse the WAV file

      sample_count = channel_count = sample_size = channel_size = 0;

      current_pos = 0;

      for (;;)

      {

         #define READ_DATA(type) *((type *) &soundfile.data[current_pos]); current_pos += sizeof (type); if (current_pos >= soundfile.size) break;

         chunk_id = READ_DATA (uint32_t);

         if (chunk_id == *((uint32_t *) "RIFF"))

         {

            temp32 = READ_DATA (uint32_t); // skip the "chunk size" field

            temp32 = READ_DATA (uint32_t); // skip the "riff style" field (typically "WAVE")

         }

         else if (chunk_id == *((uint32_t *) "fmt "))

         {

            blksiz = READ_DATA (uint32_t);

            temp16 = READ_DATA (uint16_t); if (temp16 != 1) break; // compressed WAVs are unsupported

            temp32 = READ_DATA (uint16_t); channel_count = (int) temp16;

            temp32 = READ_DATA (uint32_t); sample_rate = (int) temp32;

            temp32 = READ_DATA (uint32_t);

            temp16 = READ_DATA (uint16_t); sample_size = (int) temp16;

            temp16 = READ_DATA (uint16_t); channel_size = (int) temp16 / 8;

            if (blksiz > 16)

               current_pos += blksiz - 16;

            if (current_pos >= soundfile.size)

               break; // don't go beyond the end of the file

         }

         else if (chunk_id == *((uint32_t *) "data"))

         {

            temp32 = READ_DATA (uint32_t); sample_count = (int) temp32;

            break; // current_pos is now at the beginning of data, and data measures sample_count bytes long

         }

         else

         {

            blksiz = READ_DATA (uint32_t); // skip the "chunk size" field

            current_pos += blksiz; // useless chunk, skip it

            if (current_pos >= soundfile.size)

               break; // don't go beyond the end of the file

         }

         #undef READ_DATA

      }

      if ((sample_count == 0) || (channel_count == 0) || (sample_size == 0) || (channel_size == 0))

         return; // FIXME: not a wav file

      // compute the total number of samples (number of channels * number of frames)

      sample_count /= sample_size;

      // allocate space for one more sound buffer, and tie it to the linked list

      if (soundbuffers == NULL)

      {

         soundbuffers = (openal_buffer_t *) SAFE_malloc (1, sizeof (openal_buffer_t), false); // allocate a new slot to initialize the linked list

         soundbuffer = soundbuffers; // and set the pointer to the beginning of the list

      }

      else

      {

         for (soundbuffer = soundbuffers; soundbuffer->next != NULL; soundbuffer = soundbuffer->next); // locate the last slot in the linked list

         soundbuffer->next = (openal_buffer_t *) SAFE_malloc (1, sizeof (openal_buffer_t), false); // allocate a new slot and tie it there at the same time

         soundbuffer = soundbuffer->next; // and update the pointer to it

      }

      soundbuffer->next = NULL; // this is the last sound buffer in the linked list, do remember it.

      temp32 = wcslen (soundfile_path) + 1; // compute pathname buffer length (including null terminator)

      soundbuffer->pathname = (wchar_t *) SAFE_malloc (temp32, sizeof (wchar_t), false); // allocate space for it

      wcscpy_s (soundbuffer->pathname, temp32, soundfile_path); // and copy this sound's pathname in the newly allocated buffer

      // allocate the samples buffer and fill it, mixing all the WAV channels altogether in a 16-bit mono stream

      soundbuffer->openal_samples = (ALshort *) SAFE_malloc (sample_count, sizeof (ALshort), false);

      for (sample_index = 0; sample_index < sample_count; sample_index++)

      {

         sample_value = 0;

         if (channel_size == 1)

            for (channel_index = 0; channel_index < channel_count; channel_index++)

               sample_value += *((int8_t *) soundfile.data[current_pos + sample_index * sample_size + channel_index * channel_size]);

         else if (channel_size == 2)

            for (channel_index = 0; channel_index < channel_count; channel_index++)

               sample_value += *((int16_t *) &soundfile.data[current_pos + sample_index * sample_size + channel_index * channel_size]);

         sample_value /= channel_count;

         soundbuffer->openal_samples[sample_index] = (ALshort) sample_value;

      }

      Buffer_Forget (&soundfile); // we can now forget this sound file

      alGenBuffers (1, &soundbuffer->openal_buffer); // create an OpenAL sound buffer and fill it with our samples

      alBufferData (soundbuffer->openal_buffer, AL_FORMAT_MONO16, soundbuffer->openal_samples, sample_count * sizeof (ALushort), (ALsizei) sample_rate);

      if (alGetError () != AL_NO_ERROR)

         return; // FIXME: couldn't fill OpenAL buffer

   }

   // now we have a buffer to play

   // cycle through our known OpenAL sources and find a free one

   for (source_index = 0; source_index < source_count; source_index++)

      if (!sources[source_index].is_used)

         break; // break as soon as we find it

   // have we NOT found any ? if so, reallocate so as to have one more

   if (source_index == source_count)

   {

      sources = (openal_source_t *) SAFE_realloc (sources, source_count, source_count + 1, sizeof (openal_source_t), false);

      source_count++; // one more source has been created

   }

   // now we have a source to play our buffer

   sources[source_index].is_used = true; // immediately mark it as used

   alGenSources (1, &sources[source_index].openal_source); // (re)create an OpenAL source

   alSourcef (sources[source_index].openal_source, AL_PITCH, (ALfloat) pitch); // set the source's pitch

   alSourcef (sources[source_index].openal_source, AL_GAIN, 1.0f); // set the source's volume (full)

   alSource3f (sources[source_index].openal_source, AL_POSITION, (ALfloat) pos_x * ATTENUATION_FACTOR, (ALfloat) pos_y * ATTENUATION_FACTOR, (ALfloat) pos_z * ATTENUATION_FACTOR);

   alSource3f (sources[source_index].openal_source, AL_VELOCITY, 0, 0, 0); // set the source's velocity (static)

   alSourcei (sources[source_index].openal_source, AL_LOOPING, AL_FALSE); // set it as non-looping

   alSourcei (sources[source_index].openal_source, AL_BUFFER, soundbuffer->openal_buffer); // attach our bufferized data to it

   // play the source! Audio_Think() will dispose of it when it's finished

   alSourcePlay (sources[source_index].openal_source);

   return; // finished

}