path: root/src/i_sdlsound.c

// Emacs style mode select   -*- C++ -*- 
//-----------------------------------------------------------------------------
//
// Copyright(C) 1993-1996 Id Software, Inc.
// Copyright(C) 2005 Simon Howard
// Copyright(C) 2008 David Flater
//
// This program is free software; you can redistribute it and/or
// modify it under the terms of the GNU General Public License
// as published by the Free Software Foundation; either version 2
// of the License, or (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA
// 02111-1307, USA.
//
// DESCRIPTION:
//	System interface for sound.
//
//-----------------------------------------------------------------------------


#include "config.h"

#include <stdio.h>
#include <stdlib.h>
#include <assert.h>
#include "SDL.h"
#include "SDL_mixer.h"

#ifdef HAVE_LIBSAMPLERATE
#include <samplerate.h>
#endif

#include "deh_main.h"
#include "i_system.h"
#include "s_sound.h"
#include "m_argv.h"
#include "w_wad.h"
#include "z_zone.h"

#include "doomdef.h"

#define LOW_PASS_FILTER
#define NUM_CHANNELS 16

static boolean sound_initialised = false;

static Mix_Chunk sound_chunks[NUMSFX];
static int channels_playing[NUM_CHANNELS];

static int mixer_freq;
static Uint16 mixer_format;
static int mixer_channels;
static uint32_t (*ExpandSoundData)(byte *data, int samplerate, int length,
                                   Mix_Chunk *destination) = NULL;

int use_libsamplerate = 0;

// When a sound stops, check if it is still playing.  If it is not, 
// we can mark the sound data as CACHE to be freed back for other
// means.

static void ReleaseSoundOnChannel(int channel)
{
    int i;
    int id = channels_playing[channel];

    if (!id)
    {
        return;
    }

    channels_playing[channel] = sfx_None;
    
    for (i=0; i<NUM_CHANNELS; ++i)
    {
        // Playing on this channel? if so, don't release.

        if (channels_playing[i] == id)
            return;
    }

    // Not used on any channel, and can be safely released
    
    Z_ChangeTag(sound_chunks[id].abuf, PU_CACHE);
}

#ifdef HAVE_LIBSAMPLERATE

// Returns the conversion mode for libsamplerate to use.

static int SRC_ConversionMode(void)
{
    switch (use_libsamplerate)
    {
        // 0 = disabled

        default:
        case 0:
            return -1;

        // Ascending numbers give higher quality

        case 1:
            return SRC_LINEAR;
        case 2:
            return SRC_ZERO_ORDER_HOLD;
        case 3:
            return SRC_SINC_FASTEST;
        case 4:
            return SRC_SINC_MEDIUM_QUALITY;
        case 5:
            return SRC_SINC_BEST_QUALITY;
    }
}

// libsamplerate-based generic sound expansion function for any sample rate
//   unsigned 8 bits --> signed 16 bits
//   mono --> stereo
//   samplerate --> mixer_freq
// Returns number of clipped samples.
// DWF 2008-02-10 with cleanups by Simon Howard.

static uint32_t ExpandSoundData_SRC(byte *data,
                                    int samplerate,
                                    int length,
                                    Mix_Chunk *destination)
{
    SRC_DATA src_data;
    uint32_t i, abuf_index=0, clipped=0;
    int retn;
    int16_t *expanded;

    src_data.input_frames = length;
    src_data.data_in = malloc(length * sizeof(float));
    src_data.src_ratio = (double)mixer_freq / samplerate;

    // We include some extra space here in case of rounding-up.
    src_data.output_frames = src_data.src_ratio * length + (mixer_freq / 4);
    src_data.data_out = malloc(src_data.output_frames * sizeof(float));

    assert(src_data.data_in != NULL && src_data.data_out != NULL);

    // Convert input data to floats

    for (i=0; i<length; ++i)
    {
        // Unclear whether 128 should be interpreted as "zero" or whether a
        // symmetrical range should be assumed.  The following assumes a
        // symmetrical range.
        src_data.data_in[i] = data[i] / 127.5 - 1;
    }

    // Do the sound conversion

    retn = src_simple(&src_data, SRC_ConversionMode(), 1);
    assert(retn == 0);

    // Convert the result back into 16-bit integers.

    destination->alen = src_data.output_frames_gen * 4;
    destination->abuf = Z_Malloc(destination->alen, PU_STATIC, 
                                 &destination->abuf);
    expanded = (int16_t *) destination->abuf;

    for (i=0; i<src_data.output_frames_gen; ++i)
    {
        // libsamplerate does not limit itself to the -1.0 .. 1.0 range on
        // output, so a multiplier less than INT16_MAX (32767) is required
        // to avoid overflows or clipping.  However, the smaller the
        // multiplier, the quieter the sound effects get, and the more you
        // have to turn down the music to keep it in balance.

        // 22265 is the largest multiplier that can be used to resample all
        // of the Vanilla DOOM sound effects to 48 kHz without clipping
        // using SRC_SINC_BEST_QUALITY.  It is close enough (only slightly
        // too conservative) for SRC_SINC_MEDIUM_QUALITY and
        // SRC_SINC_FASTEST.  PWADs with interestingly different sound
        // effects or target rates other than 48 kHz might still result in
        // clipping--I don't know if there's a limit to it.

        // As the number of clipped samples increases, the signal is
        // gradually overtaken by noise, with the loudest parts going first.
        // However, a moderate amount of clipping is often tolerated in the
        // quest for the loudest possible sound overall.  The results of
        // using INT16_MAX as the multiplier are not all that bad, but
        // artifacts are noticeable during the loudest parts.

        float   cvtval_f = src_data.data_out[i] * 22265;
        int32_t cvtval_i = cvtval_f + (cvtval_f < 0 ? -0.5 : 0.5);

        // Asymmetrical sound worries me, so we won't use -32768.
        if (cvtval_i < -INT16_MAX) {
          cvtval_i = -INT16_MAX;
          ++clipped;
        } else if (cvtval_i > INT16_MAX) {
          cvtval_i = INT16_MAX;
          ++clipped;
        }

        // Left and right channels

        expanded[abuf_index++] = cvtval_i;
        expanded[abuf_index++] = cvtval_i;
    }

    free(src_data.data_in);
    free(src_data.data_out);
    return clipped;
}

#endif

static boolean ConvertibleRatio(int freq1, int freq2)
{
    int ratio;

    if (freq1 > freq2)
    {
        return ConvertibleRatio(freq2, freq1);
    }
    else if ((freq2 % freq1) != 0)
    {
        // Not in a direct ratio

        return false;
    }
    else
    {
        // Check the ratio is a power of 2

        ratio = freq2 / freq1;

        while ((ratio & 1) == 0)
        {
            ratio = ratio >> 1;
        }

        return ratio == 1;
    }
}

// Generic sound expansion function for any sample rate.
// Returns number of clipped samples (always 0).

static uint32_t ExpandSoundData_SDL(byte *data,
                                    int samplerate,
                                    int length,
                                    Mix_Chunk *destination)
{
    SDL_AudioCVT convertor;
    uint32_t expanded_length;
 
    // Calculate the length of the expanded version of the sample.    

    expanded_length = (uint32_t) ((((uint64_t) length) * mixer_freq) / samplerate);

    // Double up twice: 8 -> 16 bit and mono -> stereo

    expanded_length *= 4;

    destination->alen = expanded_length;
    destination->abuf 
        = Z_Malloc(expanded_length, PU_STATIC, &destination->abuf);

    // If we can, use the standard / optimised SDL conversion routines.
    
    if (samplerate <= mixer_freq
     && ConvertibleRatio(samplerate, mixer_freq)
     && SDL_BuildAudioCVT(&convertor,
                          AUDIO_U8, 1, samplerate,
                          mixer_format, mixer_channels, mixer_freq))
    {
        convertor.buf = destination->abuf;
        convertor.len = length;
        memcpy(convertor.buf, data, length);

        SDL_ConvertAudio(&convertor);
    }
    else
    {
        Sint16 *expanded = (Sint16 *) destination->abuf;
        int expanded_length;
        int expand_ratio;
        int i;

        // Generic expansion if conversion does not work:
        //
        // SDL's audio conversion only works for rate conversions that are
        // powers of 2; if the two formats are not in a direct power of 2
        // ratio, do this naive conversion instead.

        // number of samples in the converted sound

        expanded_length = ((uint64_t) length * mixer_freq) / samplerate;
        expand_ratio = (length << 8) / expanded_length;

        for (i=0; i<expanded_length; ++i)
        {
            Sint16 sample;
            int src;

            src = (i * expand_ratio) >> 8;

            sample = data[src] | (data[src] << 8);
            sample -= 32768;

            // expand 8->16 bits, mono->stereo

            expanded[i * 2] = expanded[i * 2 + 1] = sample;
        }

#ifdef LOW_PASS_FILTER
        // Perform a low-pass filter on the upscaled sound to filter
        // out high-frequency noise from the conversion process.

        {
            float rc, dt, alpha;

            // Low-pass filter for cutoff frequency f:
            //
            // For sampling rate r, dt = 1 / r
            // rc = 1 / 2*pi*f
            // alpha = dt / (rc + dt)

            // Filter to the half sample rate of the original sound effect
            // (maximum frequency, by nyquist)

            dt = 1.0f / mixer_freq;
            rc = 1.0f / (3.14f * samplerate);
            alpha = dt / (rc + dt);

            for (i=1; i<expanded_length; ++i) 
            {
                expanded[i] = (Sint16) (alpha * expanded[i] + (1 - alpha) * expanded[i-1]);
            }
        }
#endif /* #ifdef LOW_PASS_FILTER */
    }

    return 0;
}

// Load and convert a sound effect
// Returns true if successful

static boolean CacheSFX(int sound)
{
    int lumpnum;
    unsigned int lumplen;
    int samplerate;
    int clipped;
    unsigned int length;
    byte *data;

    // need to load the sound

    lumpnum = S_sfx[sound].lumpnum;
    data = W_CacheLumpNum(lumpnum, PU_STATIC);
    lumplen = W_LumpLength(lumpnum);

    // Check the header, and ensure this is a valid sound

    if (lumplen < 8
     || data[0] != 0x03 || data[1] != 0x00)
    {
        // Invalid sound

        return false;
    }

    // 16 bit sample rate field, 32 bit length field

    samplerate = (data[3] << 8) | data[2];
    length = (data[7] << 24) | (data[6] << 16) | (data[5] << 8) | data[4];

    // If the header specifies that the length of the sound is greater than
    // the length of the lump itself, this is an invalid sound lump

    if (length > lumplen - 8)
    {
        return false;
    }

    // Sample rate conversion
    // DWF 2008-02-10:  sound_chunks[sound].alen and abuf are determined
    // by ExpandSoundData.

    sound_chunks[sound].allocated = 1;
    sound_chunks[sound].volume = MIX_MAX_VOLUME;

    clipped = ExpandSoundData(data + 8, 
                              samplerate, 
                              length, 
                              &sound_chunks[sound]);

    if (clipped)
    {
        fprintf(stderr, "Sound %d: clipped %u samples (%0.2f %%)\n", 
                        sound, clipped,
                        400.0 * clipped / sound_chunks[sound].alen);
    }

    // don't need the original lump any more
  
    W_ReleaseLumpNum(lumpnum);

    return true;
}

#ifdef HAVE_LIBSAMPLERATE

// Preload all the sound effects - stops nasty ingame freezes

static void I_PrecacheSounds(void)
{
    char namebuf[9];
    int i;

    printf("I_PrecacheSounds: Precaching all sound effects..");

    for (i=sfx_pistol; i<NUMSFX; ++i)
    {
        if ((i % 6) == 0)
        {
            printf(".");
            fflush(stdout);
        }

        sprintf(namebuf, "ds%s", DEH_String(S_sfx[i].name));

        S_sfx[i].lumpnum = W_CheckNumForName(namebuf);

        if (S_sfx[i].lumpnum != -1)
        {
            CacheSFX(i);

            if (sound_chunks[i].abuf != NULL)
            {
                Z_ChangeTag(sound_chunks[i].abuf, PU_CACHE);
            }
        }
    }

    printf("\n");
}

#endif

static Mix_Chunk *GetSFXChunk(int sound_id)
{
    if (sound_chunks[sound_id].abuf == NULL)
    {
        if (!CacheSFX(sound_id))
            return NULL;
    }
    else
    {
        // don't free the sound while it is playing!
   
        Z_ChangeTag(sound_chunks[sound_id].abuf, PU_STATIC);
    }

    return &sound_chunks[sound_id];
}


//
// Retrieve the raw data lump index
//  for a given SFX name.
//

static int I_SDL_GetSfxLumpNum(sfxinfo_t* sfx)
{
    char namebuf[9];

    sprintf(namebuf, "ds%s", DEH_String(sfx->name));
    
    return W_GetNumForName(namebuf);
}

static void I_SDL_UpdateSoundParams(int handle, int vol, int sep)
{
    int left, right;

    if (!sound_initialised)
    {
        return;
    }

    left = ((254 - sep) * vol) / 127;
    right = ((sep) * vol) / 127;

    Mix_SetPanning(handle, left, right);
}


//
// Starting a sound means adding it
//  to the current list of active sounds
//  in the internal channels.
// As the SFX info struct contains
//  e.g. a pointer to the raw data,
//  it is ignored.
// As our sound handling does not handle
//  priority, it is ignored.
// Pitching (that is, increased speed of playback)
//  is set, but currently not used by mixing.
//

static int I_SDL_StartSound(int id, int channel, int vol, int sep)
{
    Mix_Chunk *chunk;

    if (!sound_initialised)
    {
        return -1;
    }

    // Release a sound effect if there is already one playing
    // on this channel

    ReleaseSoundOnChannel(channel);

    // Get the sound data

    chunk = GetSFXChunk(id);

    if (chunk == NULL)
    {
        return -1;
    }

    // play sound

    Mix_PlayChannelTimed(channel, chunk, 0, -1);

    channels_playing[channel] = id;

    // set separation, etc.
 
    I_SDL_UpdateSoundParams(channel, vol, sep);

    return channel;
}

static void I_SDL_StopSound (int handle)
{
    if (!sound_initialised)
    {
        return;
    }

    Mix_HaltChannel(handle);

    // Sound data is no longer needed; release the
    // sound data being used for this channel

    ReleaseSoundOnChannel(handle);
}


static boolean I_SDL_SoundIsPlaying(int handle)
{
    if (handle < 0)
    {
        return false;
    }

    return Mix_Playing(handle);
}

// 
// Periodically called to update the sound system
//

static void I_SDL_UpdateSound(void)
{
    int i;

    // Check all channels to see if a sound has finished

    for (i=0; i<NUM_CHANNELS; ++i)
    {
        if (channels_playing[i] && !I_SDL_SoundIsPlaying(i))
        {
            // Sound has finished playing on this channel,
            // but sound data has not been released to cache
            
            ReleaseSoundOnChannel(i);
        }
    }
}

static void I_SDL_ShutdownSound(void)
{    
    if (!sound_initialised)
    {
        return;
    }

    Mix_CloseAudio();
    SDL_QuitSubSystem(SDL_INIT_AUDIO);

    sound_initialised = false;
}


static boolean I_SDL_InitSound(void)
{ 
    int i;
    
    // No sounds yet

    for (i=0; i<NUMSFX; ++i)
    {
        sound_chunks[i].abuf = NULL;
    }

    for (i=0; i<NUM_CHANNELS; ++i)
    {
        channels_playing[i] = sfx_None;
    }

    if (SDL_Init(SDL_INIT_AUDIO) < 0)
    {
        fprintf(stderr, "Unable to set up sound.\n");
        return false;
    }

    if (Mix_OpenAudio(snd_samplerate, AUDIO_S16SYS, 2, 1024) < 0)
    {
        fprintf(stderr, "Error initialising SDL_mixer: %s\n", Mix_GetError());
        return false;
    }

    ExpandSoundData = ExpandSoundData_SDL;

    Mix_QuerySpec(&mixer_freq, &mixer_format, &mixer_channels);

#ifdef HAVE_LIBSAMPLERATE
    if (use_libsamplerate != 0)
    {
        if (SRC_ConversionMode() < 0)
        {
            I_Error("I_SDL_InitSound: Invalid value for use_libsamplerate: %i",
                    use_libsamplerate);
        }

        ExpandSoundData = ExpandSoundData_SRC;

        I_PrecacheSounds();
    }
#else
    if (use_libsamplerate != 0)
    {
        fprintf(stderr, "I_SDL_InitSound: use_libsamplerate=%i, but "
                        "libsamplerate support not compiled in.\n",
                        use_libsamplerate);
    }
#endif

    Mix_AllocateChannels(NUM_CHANNELS);
    
    SDL_PauseAudio(0);

    sound_initialised = true;

    return true;
}

static snddevice_t sound_sdl_devices[] = 
{
    SNDDEVICE_SB,
    SNDDEVICE_PAS,
    SNDDEVICE_GUS,
    SNDDEVICE_WAVEBLASTER,
    SNDDEVICE_SOUNDCANVAS,
    SNDDEVICE_AWE32,
};

sound_module_t sound_sdl_module = 
{
    sound_sdl_devices,
    arrlen(sound_sdl_devices),
    I_SDL_InitSound,
    I_SDL_ShutdownSound,
    I_SDL_GetSfxLumpNum,
    I_SDL_UpdateSound,
    I_SDL_UpdateSoundParams,
    I_SDL_StartSound,
    I_SDL_StopSound,
    I_SDL_SoundIsPlaying,
};