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https://github.com/libretro/scummvm.git
synced 2025-01-22 10:17:22 +00:00
Added infrastructure to support more than 8 bit signed / unsigned RAW
mixers. Porters, beware, the default configuration is now 16 bit stereo instead of 16 mono as before (I changed X11 and SDL but no others). I did not add support for any other format yet, I will let Endy do it when he needs it :-) svn-id: r4348
This commit is contained in:
Lionel Ulmer
parent
53f993be44
commit
3b4c6ceb0f
2
sdl.cpp
2
sdl.cpp
@ -829,7 +829,7 @@ bool OSystem_SDL::set_sound_proc(void *param, SoundProc *proc, byte format) {
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desired.freq = SAMPLES_PER_SEC;
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desired.format = AUDIO_S16SYS;
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desired.channels = 1;
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desired.channels = 2;
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desired.samples = 2048;
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desired.callback = proc;
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desired.userdata = param;
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@ -84,14 +84,20 @@ int SoundMixer::play_mp3_cdtrack(PlayingSoundHandle *handle, FILE* file, mad_tim
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void SoundMixer::mix(int16 *buf, uint len) {
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if (_paused)
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if (_paused) {
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memset(buf, 0, 2 * len * sizeof(int16));
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return;
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}
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if (_premix_proc) {
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int i;
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_premix_proc(_premix_param, buf, len);
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for (i = (len - 1); i >= 0; i--) {
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buf[2 * i] = buf[2 * i + 1] = buf[i];
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}
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} else {
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/* no premixer available, zero the buf out */
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memset(buf, 0, len * sizeof(int16));
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memset(buf, 0, 2 * len * sizeof(int16));
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}
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/* now mix all channels */
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@ -101,7 +107,7 @@ void SoundMixer::mix(int16 *buf, uint len) {
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}
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void SoundMixer::on_generate_samples(void *s, byte *samples, int len) {
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((SoundMixer*)s)->mix((int16*)samples, len>>1);
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((SoundMixer*)s)->mix((int16*)samples, len>>2);
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}
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bool SoundMixer::bind_to_system(OSystem *syst) {
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@ -207,6 +213,64 @@ void SoundMixer::Channel_RAW::append(void *data, uint32 len) {
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_mixer->_paused = false; /* Mix again now */
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}
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static void mix_signed_mono_8(int16 *data, uint len, byte **s_ptr, uint32 *fp_pos_ptr, int fp_speed, const int16 *vol_tab) {
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uint32 fp_pos = *fp_pos_ptr;
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byte *s = *s_ptr;
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do {
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fp_pos += fp_speed;
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*data++ += vol_tab[*s];
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*data++ += vol_tab[*s];
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s += fp_pos >> 16;
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fp_pos &= 0x0000FFFF;
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} while (--len);
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*fp_pos_ptr = fp_pos;
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*s_ptr = s;
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}
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static void mix_unsigned_mono_8(int16 *data, uint len, byte **s_ptr, uint32 *fp_pos_ptr, int fp_speed, const int16 *vol_tab) {
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uint32 fp_pos = *fp_pos_ptr;
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byte *s = *s_ptr;
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do {
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fp_pos += fp_speed;
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*data++ += vol_tab[*s ^ 0x80];
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*data++ += vol_tab[*s ^ 0x80];
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s += fp_pos >> 16;
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fp_pos &= 0x0000FFFF;
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} while (--len);
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*fp_pos_ptr = fp_pos;
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*s_ptr = s;
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}
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static void mix_signed_stereo_8(int16 *data, uint len, byte **s_ptr, uint32 *fp_pos_ptr, int fp_speed, const int16 *vol_tab) {
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warning("Mixing stereo signed 8 bit is not supported yet ");
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}
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static void mix_unsigned_stereo_8(int16 *data, uint len, byte **s_ptr, uint32 *fp_pos_ptr, int fp_speed, const int16 *vol_tab) {
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warning("Mixing stereo unsigned 8 bit is not supported yet ");
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}
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static void mix_signed_mono_16(int16 *data, uint len, byte **s_ptr, uint32 *fp_pos_ptr, int fp_speed, const int16 *vol_tab) {
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warning("Mixing mono signed 16 bit is not supported yet ");
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}
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static void mix_unsigned_mono_16(int16 *data, uint len, byte **s_ptr, uint32 *fp_pos_ptr, int fp_speed, const int16 *vol_tab) {
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warning("Mixing mono unsigned 16 bit is not supported yet ");
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}
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static void mix_signed_stereo_16(int16 *data, uint len, byte **s_ptr, uint32 *fp_pos_ptr, int fp_speed, const int16 *vol_tab) {
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warning("Mixing stereo signed 16 bit is not supported yet ");
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}
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static void mix_unsigned_stereo_16(int16 *data, uint len, byte **s_ptr, uint32 *fp_pos_ptr, int fp_speed, const int16 *vol_tab) {
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warning("Mixing stereo unsigned 16 bit is not supported yet ");
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}
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static void (*mixer_helper_table[16])(int16 *data, uint len, byte **s_ptr, uint32 *fp_pos_ptr, int fp_speed, const int16 *vol_tab) = {
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mix_signed_mono_8,
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mix_unsigned_mono_8,
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mix_signed_stereo_8,
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mix_unsigned_stereo_8,
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mix_signed_mono_16,
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mix_unsigned_mono_16,
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mix_signed_stereo_16,
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mix_unsigned_stereo_16
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};
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void SoundMixer::Channel_RAW::mix(int16 *data, uint len) {
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byte *s, *s_org = NULL;
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uint32 fp_pos;
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@ -245,21 +309,7 @@ void SoundMixer::Channel_RAW::mix(int16 *data, uint len) {
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const uint32 fp_speed = _fp_speed;
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const int16 *vol_tab = _mixer->_volume_table;
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if (_flags & FLAG_UNSIGNED) {
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do {
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fp_pos += fp_speed;
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*data++ += vol_tab[*s ^ 0x80];
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s += fp_pos >> 16;
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fp_pos &= 0x0000FFFF;
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} while (--len);
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} else {
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do {
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fp_pos += fp_speed;
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*data++ += vol_tab[*s];
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s += fp_pos >> 16;
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fp_pos &= 0x0000FFFF;
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} while (--len);
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}
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mixer_helper_table[_flags & 0x07](data, len, &s, &fp_pos, fp_speed, vol_tab);
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_pos = s - (byte*) _ptr;
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_fp_pos = fp_pos;
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@ -345,7 +395,9 @@ void SoundMixer::Channel_MP3::mix(int16 *data, uint len) {
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if (_silence_cut > 0) {
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_silence_cut--;
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} else {
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*data++ += (int16) ((scale_sample(*ch++) * volume) / 32);
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int16 sample = (int16) ((scale_sample(*ch++) * volume) / 32);
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*data++ += sample;
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*data++ += sample;
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len--;
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}
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_pos_in_frame++;
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@ -463,7 +515,9 @@ void SoundMixer::Channel_MP3_CDMUSIC::mix(int16 *data, uint len) {
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// Get samples, play samples ...
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ch = _synth.pcm.samples[0] + _pos_in_frame;
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while ((_pos_in_frame < _synth.pcm.length) && (len > 0)) {
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*data++ += (int16) ((scale_sample(*ch++) * volume) / 32);
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int16 sample = (int16) ((scale_sample(*ch++) * volume) / 32);
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*data++ += sample;
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*data++ += sample;
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len--;
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_pos_in_frame++;
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}
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@ -131,9 +131,12 @@ public:
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/* start playing a raw sound */
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enum {
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FLAG_AUTOFREE = 1,
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FLAG_UNSIGNED = 2, /* unsigned samples */
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FLAG_FILE = 4, /* sound is a FILE * that's read from */
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/* Do *NOT* change any of these flags without looking at the code in mixer.cpp */
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FLAG_UNSIGNED = 1, /* unsigned samples */
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FLAG_STEREO = 2, /* sound is in stereo */
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FLAG_16BITS = 4, /* sound is 16 bits wide */
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FLAG_AUTOFREE = 8, /* sound buffer is freed automagically at the end of playing */
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FLAG_FILE = 16, /* sound is a FILE * that's read from */
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};
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int play_raw(PlayingSoundHandle *handle, void *sound, uint32 size, uint rate, byte flags);
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#ifdef COMPRESSED_SOUND_FILE
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11
x11.cpp
11
x11.cpp
@ -256,21 +256,18 @@ static void *sound_and_music_thread(void *params)
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sched_yield();
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while (1) {
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unsigned short *buf = (unsigned short *)sound_buffer;
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unsigned char *buf = (unsigned char *)sound_buffer;
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int size, written;
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sound_proc(proc_param, (byte *)sound_buffer, FRAG_SIZE >> 1);
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/* Now convert to stereo */
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for (int i = ((FRAG_SIZE >> 2) - 1); i >= 0; i--) {
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buf[2 * i + 1] = buf[2 * i] = buf[i];
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}
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sound_proc(proc_param, (byte *)sound_buffer, FRAG_SIZE);
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#ifdef CAPTURE_SOUND
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fwrite(buf, 2, FRAG_SIZE >> 1, f);
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fflush(f);
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#endif
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size = FRAG_SIZE;
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while (size > 0) {
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written = write(sound_fd, sound_buffer, size);
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written = write(sound_fd, buf, size);
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buf += written;
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size -= written;
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}
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}
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