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authorrubidium <rubidium@openttd.org>2009-08-09 23:04:08 +0000
committerrubidium <rubidium@openttd.org>2009-08-09 23:04:08 +0000
commit8e058c2d27961bdd1db883679b5743f418e8b854 (patch)
tree2f662623f2b81ae0cf44c9bb9366ac62139dd118 /src
parent9ee2a66c8620d2f47c0d7792847c90146dfc4f8e (diff)
downloadopenttd-8e058c2d27961bdd1db883679b5743f418e8b854.tar.xz
(svn r17140) -Change: allow higher sample rate and higher quality samples. Based on work by orudge.
Diffstat (limited to 'src')
-rw-r--r--src/mixer.cpp55
-rw-r--r--src/mixer.h2
-rw-r--r--src/sound.cpp34
-rw-r--r--src/sound/allegro_s.cpp2
-rw-r--r--src/sound/cocoa_s.cpp2
-rw-r--r--src/sound/sdl_s.cpp2
-rw-r--r--src/sound/win32_s.cpp4
7 files changed, 80 insertions, 21 deletions
diff --git a/src/mixer.cpp b/src/mixer.cpp
index cf8b30c9a..5a8370df8 100644
--- a/src/mixer.cpp
+++ b/src/mixer.cpp
@@ -21,6 +21,8 @@ struct MixerChannel {
/* Mixing volume */
int volume_left;
int volume_right;
+
+ bool is16bit;
};
static MixerChannel _channels[8];
@@ -34,6 +36,46 @@ static uint32 _play_rate = 11025;
*/
static const int MAX_VOLUME = 128 * 128;
+static void mix_int16(MixerChannel *sc, int16 *buffer, uint samples)
+{
+ int16 *b;
+ uint32 frac_pos;
+ uint32 frac_speed;
+ int volume_left;
+ int volume_right;
+
+ if (samples > sc->samples_left) samples = sc->samples_left;
+ sc->samples_left -= samples;
+ assert(samples > 0);
+
+ b = (int16*)sc->memory + sc->pos;
+ frac_pos = sc->frac_pos;
+ frac_speed = sc->frac_speed;
+ volume_left = sc->volume_left;
+ volume_right = sc->volume_right;
+
+ if (frac_speed == 0x10000) {
+ /* Special case when frac_speed is 0x10000 */
+ do {
+ buffer[0] = Clamp(buffer[0] + (*b * volume_left >> 16), -MAX_VOLUME, MAX_VOLUME);
+ buffer[1] = Clamp(buffer[1] + (*b * volume_right >> 16), -MAX_VOLUME, MAX_VOLUME);
+ b++;
+ buffer += 2;
+ } while (--samples > 0);
+ } else {
+ do {
+ buffer[0] = Clamp(buffer[0] + (*b * volume_left >> 16), -MAX_VOLUME, MAX_VOLUME);
+ buffer[1] = Clamp(buffer[1] + (*b * volume_right >> 16), -MAX_VOLUME, MAX_VOLUME);
+ buffer += 2;
+ frac_pos += frac_speed;
+ b += frac_pos >> 16;
+ frac_pos &= 0xffff;
+ } while (--samples > 0);
+ }
+
+ sc->frac_pos = frac_pos;
+ sc->pos = b - (int16*)sc->memory;
+}
static void mix_int8_to_int16(MixerChannel *sc, int16 *buffer, uint samples)
{
@@ -93,7 +135,11 @@ void MxMixSamples(void *buffer, uint samples)
/* Mix each channel */
for (mc = _channels; mc != endof(_channels); mc++) {
if (mc->active) {
- mix_int8_to_int16(mc, (int16*)buffer, samples);
+ if (mc->is16bit) {
+ mix_int16(mc, (int16*)buffer, samples);
+ } else {
+ mix_int8_to_int16(mc, (int16*)buffer, samples);
+ }
if (mc->samples_left == 0) MxCloseChannel(mc);
}
}
@@ -110,7 +156,7 @@ MixerChannel *MxAllocateChannel()
return NULL;
}
-void MxSetChannelRawSrc(MixerChannel *mc, int8 *mem, size_t size, uint rate)
+void MxSetChannelRawSrc(MixerChannel *mc, int8 *mem, size_t size, uint rate, bool is16bit)
{
mc->memory = mem;
mc->frac_pos = 0;
@@ -124,7 +170,10 @@ void MxSetChannelRawSrc(MixerChannel *mc, int8 *mem, size_t size, uint rate)
rate = (rate >> 1) + 1;
}
- mc->samples_left = (uint)size * _play_rate / rate;
+ int div = is16bit ? 2 : 1;
+
+ mc->samples_left = (uint)size * _play_rate / rate / div;
+ mc->is16bit = is16bit;
}
void MxSetChannelVolume(MixerChannel *mc, uint left, uint right)
diff --git a/src/mixer.h b/src/mixer.h
index cdd6f65b4..fcb3faade 100644
--- a/src/mixer.h
+++ b/src/mixer.h
@@ -11,7 +11,7 @@ bool MxInitialize(uint rate);
void MxMixSamples(void *buffer, uint samples);
MixerChannel *MxAllocateChannel();
-void MxSetChannelRawSrc(MixerChannel *mc, int8 *mem, size_t size, uint rate);
+void MxSetChannelRawSrc(MixerChannel *mc, int8 *mem, size_t size, uint rate, bool is16bit);
void MxSetChannelVolume(MixerChannel *mc, uint left, uint right);
void MxActivateChannel(MixerChannel*);
diff --git a/src/sound.cpp b/src/sound.cpp
index 99799002d..c1cb1c02e 100644
--- a/src/sound.cpp
+++ b/src/sound.cpp
@@ -29,7 +29,12 @@ static void OpenBankFile(const char *filename)
FioOpenFile(SOUND_SLOT, filename);
size_t pos = FioGetPos();
- uint count = FioReadDword() / 8;
+ uint count = FioReadDword();
+
+ /* The new format has the highest bit always set */
+ bool new_format = HasBit(count, 31);
+ ClrBit(count, 31);
+ count /= 8;
/* Simple check for the correct number of original sounds. */
if (count != ORIGINAL_SAMPLE_COUNT) {
@@ -44,7 +49,7 @@ static void OpenBankFile(const char *filename)
for (uint i = 0; i != ORIGINAL_SAMPLE_COUNT; i++) {
_original_sounds[i].file_slot = SOUND_SLOT;
- _original_sounds[i].file_offset = FioReadDword() + pos;
+ _original_sounds[i].file_offset = GB(FioReadDword(), 0, 31) + pos;
_original_sounds[i].file_size = FioReadDword();
}
@@ -56,7 +61,7 @@ static void OpenBankFile(const char *filename)
/* Check for special case, see else case */
FioReadBlock(name, FioReadByte()); // Read the name of the sound
- if (strcmp(name, "Corrupt sound") != 0) {
+ if (new_format || strcmp(name, "Corrupt sound") != 0) {
FioSeekTo(12, SEEK_CUR); // Skip past RIFF header
/* Read riff tags */
@@ -66,11 +71,11 @@ static void OpenBankFile(const char *filename)
if (tag == ' tmf') {
FioReadWord(); // wFormatTag
- sound->channels = FioReadWord(); // wChannels
- FioReadDword(); // samples per second
- sound->rate = 11025; // seems like all samples should be played at this rate.
- FioReadDword(); // avg bytes per second
- FioReadWord(); // alignment
+ sound->channels = FioReadWord(); // wChannels
+ sound->rate = FioReadDword(); // samples per second
+ if (!new_format) sound->rate = 11025; // seems like all old samples should be played at this rate.
+ FioReadDword(); // avg bytes per second
+ FioReadWord(); // alignment
sound->bits_per_sample = FioReadByte(); // bits per sample
FioSeekTo(size - (2 + 2 + 4 + 4 + 2 + 1), SEEK_CUR);
} else if (tag == 'atad') {
@@ -109,13 +114,18 @@ static bool SetBankSource(MixerChannel *mc, const SoundEntry *sound)
FioSeekToFile(sound->file_slot, sound->file_offset);
FioReadBlock(mem, sound->file_size);
- for (uint i = 0; i != sound->file_size; i++) {
- mem[i] += -128; // Convert unsigned sound data to signed
+ /* 16-bit PCM WAV files should be signed by default */
+ if (sound->bits_per_sample == 8) {
+ for (uint i = 0; i != sound->file_size; i++) {
+ mem[i] += -128; // Convert unsigned sound data to signed
+ }
}
- assert(sound->bits_per_sample == 8 && sound->channels == 1 && sound->file_size != 0 && sound->rate != 0);
+ assert(sound->bits_per_sample == 8 || sound->bits_per_sample == 16);
+ assert(sound->channels == 1);
+ assert(sound->file_size != 0 && sound->rate != 0);
- MxSetChannelRawSrc(mc, mem, sound->file_size, sound->rate);
+ MxSetChannelRawSrc(mc, mem, sound->file_size, sound->rate, sound->bits_per_sample == 16);
return true;
}
diff --git a/src/sound/allegro_s.cpp b/src/sound/allegro_s.cpp
index a94f6d238..26d184ab5 100644
--- a/src/sound/allegro_s.cpp
+++ b/src/sound/allegro_s.cpp
@@ -57,7 +57,7 @@ const char *SoundDriver_Allegro::Start(const char * const *parm)
return "No sound card found";
}
- _stream = play_audio_stream(BUFFER_SIZE, 16, true, 11025, 255, 128);
+ _stream = play_audio_stream(BUFFER_SIZE, 16, true, 44100, 255, 128);
MxInitialize(11025);
return NULL;
}
diff --git a/src/sound/cocoa_s.cpp b/src/sound/cocoa_s.cpp
index f50b6bd94..4e4fdd20a 100644
--- a/src/sound/cocoa_s.cpp
+++ b/src/sound/cocoa_s.cpp
@@ -47,7 +47,7 @@ const char *SoundDriver_Cocoa::Start(const char * const *parm)
requestedDesc.mFormatID = kAudioFormatLinearPCM;
requestedDesc.mFormatFlags = kLinearPCMFormatFlagIsPacked;
requestedDesc.mChannelsPerFrame = 2;
- requestedDesc.mSampleRate = GetDriverParamInt(parm, "hz", 11025);
+ requestedDesc.mSampleRate = GetDriverParamInt(parm, "hz", 44100);
requestedDesc.mBitsPerChannel = 16;
requestedDesc.mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
diff --git a/src/sound/sdl_s.cpp b/src/sound/sdl_s.cpp
index 7c89331f7..df7673da0 100644
--- a/src/sound/sdl_s.cpp
+++ b/src/sound/sdl_s.cpp
@@ -25,7 +25,7 @@ const char *SoundDriver_SDL::Start(const char * const *parm)
const char *s = SdlOpen(SDL_INIT_AUDIO);
if (s != NULL) return s;
- spec.freq = GetDriverParamInt(parm, "hz", 11025);
+ spec.freq = GetDriverParamInt(parm, "hz", 44100);
spec.format = AUDIO_S16SYS;
spec.channels = 2;
spec.samples = 512;
diff --git a/src/sound/win32_s.cpp b/src/sound/win32_s.cpp
index 776158adf..a7327863e 100644
--- a/src/sound/win32_s.cpp
+++ b/src/sound/win32_s.cpp
@@ -56,11 +56,11 @@ const char *SoundDriver_Win32::Start(const char * const *parm)
wfex.wFormatTag = WAVE_FORMAT_PCM;
wfex.nChannels = 2;
wfex.wBitsPerSample = 16;
- wfex.nSamplesPerSec = GetDriverParamInt(parm, "hz", 11025);
+ wfex.nSamplesPerSec = GetDriverParamInt(parm, "hz", 44100);
wfex.nBlockAlign = (wfex.nChannels * wfex.wBitsPerSample) / 8;
wfex.nAvgBytesPerSec = wfex.nSamplesPerSec * wfex.nBlockAlign;
- _bufsize = GetDriverParamInt(parm, "bufsize", (GB(GetVersion(), 0, 8) > 5) ? 2048 : 1024);
+ _bufsize = GetDriverParamInt(parm, "bufsize", (GB(GetVersion(), 0, 8) > 5) ? 8192 : 4096);
if (waveOutOpen(&_waveout, WAVE_MAPPER, &wfex, (DWORD_PTR)&waveOutProc, 0, CALLBACK_FUNCTION) != MMSYSERR_NOERROR)
return "waveOutOpen failed";