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Update: Translations from eints
luxembourgish: 3 changes by phreeze83
hebrew: 125 changes by haimlm
luxembourgish: 3 changes by phreeze83
hebrew: 125 changes by haimlm
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r14273:79f9c8f3fcad r23607:36c15679007d r5584:545d748cc681 r5584:545d748cc681 r26536:a1a59110a08b r26536:a1a59110a08b r26536:a1a59110a08b r26536:a1a59110a08b r26536:a1a59110a08b | /*
* This file is part of OpenTTD.
* OpenTTD 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, version 2.
* OpenTTD 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 OpenTTD. If not, see <http://www.gnu.org/licenses/>.
*/
/** @file mixer.cpp Mixing of sound samples. */
#include "stdafx.h"
#include <math.h>
#include <mutex>
#include <atomic>
#include "core/math_func.hpp"
#include "framerate_type.h"
#include "settings_type.h"
#include "safeguards.h"
#include "mixer.h"
struct MixerChannel {
/* pointer to allocated buffer memory */
int8 *memory;
/* current position in memory */
uint32 pos;
uint32 frac_pos;
uint32 frac_speed;
uint32 samples_left;
/* Mixing volume */
int volume_left;
int volume_right;
bool is16bit;
};
static std::atomic<uint8> _active_channels;
static MixerChannel _channels[8];
static uint32 _play_rate = 11025;
static uint32 _max_size = UINT_MAX;
static MxStreamCallback _music_stream = nullptr;
static std::mutex _music_stream_mutex;
static std::atomic<uint8> _effect_vol;
/**
* The theoretical maximum volume for a single sound sample. Multiple sound
* samples should not exceed this limit as it will sound too loud. It also
* stops overflowing when too many sounds are played at the same time, which
* causes an even worse sound quality.
*/
static const int MAX_VOLUME = 32767;
/**
* Perform the rate conversion between the input and output.
* @param b the buffer to read the data from
* @param frac_pos the position from the begin of the buffer till the next element
* @tparam T the size of the buffer (8 or 16 bits)
* @return the converted value.
*/
template <typename T>
static int RateConversion(T *b, int frac_pos)
{
return ((b[0] * ((1 << 16) - frac_pos)) + (b[1] * frac_pos)) >> 16;
}
static void mix_int16(MixerChannel *sc, int16 *buffer, uint samples, uint8 effect_vol)
{
if (samples > sc->samples_left) samples = sc->samples_left;
sc->samples_left -= samples;
assert(samples > 0);
const int16 *b = (const int16 *)sc->memory + sc->pos;
uint32 frac_pos = sc->frac_pos;
uint32 frac_speed = sc->frac_speed;
int volume_left = sc->volume_left * effect_vol / 255;
int volume_right = sc->volume_right * effect_vol / 255;
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 {
int data = RateConversion(b, frac_pos);
buffer[0] = Clamp(buffer[0] + (data * volume_left >> 16), -MAX_VOLUME, MAX_VOLUME);
buffer[1] = Clamp(buffer[1] + (data * 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 - (const int16 *)sc->memory;
}
static void mix_int8_to_int16(MixerChannel *sc, int16 *buffer, uint samples, uint8 effect_vol)
{
if (samples > sc->samples_left) samples = sc->samples_left;
sc->samples_left -= samples;
assert(samples > 0);
const int8 *b = sc->memory + sc->pos;
uint32 frac_pos = sc->frac_pos;
uint32 frac_speed = sc->frac_speed;
int volume_left = sc->volume_left * effect_vol / 255;
int volume_right = sc->volume_right * effect_vol / 255;
if (frac_speed == 0x10000) {
/* Special case when frac_speed is 0x10000 */
do {
buffer[0] = Clamp(buffer[0] + (*b * volume_left >> 8), -MAX_VOLUME, MAX_VOLUME);
buffer[1] = Clamp(buffer[1] + (*b * volume_right >> 8), -MAX_VOLUME, MAX_VOLUME);
b++;
buffer += 2;
} while (--samples > 0);
} else {
do {
int data = RateConversion(b, frac_pos);
buffer[0] = Clamp(buffer[0] + (data * volume_left >> 8), -MAX_VOLUME, MAX_VOLUME);
buffer[1] = Clamp(buffer[1] + (data * volume_right >> 8), -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 - sc->memory;
}
static void MxCloseChannel(uint8 channel_index)
{
_active_channels.fetch_and(~(1 << channel_index), std::memory_order_release);
}
void MxMixSamples(void *buffer, uint samples)
{
PerformanceMeasurer framerate(PFE_SOUND);
static uint last_samples = 0;
if (samples != last_samples) {
framerate.SetExpectedRate((double)_play_rate / samples);
last_samples = samples;
}
/* Clear the buffer */
memset(buffer, 0, sizeof(int16) * 2 * samples);
{
std::lock_guard<std::mutex> lock{ _music_stream_mutex };
/* Fetch music if a sampled stream is available */
if (_music_stream) _music_stream((int16*)buffer, samples);
}
/* Apply simple x^3 scaling to master effect volume. This increases the
* perceived difference in loudness to better match expectations. effect_vol
* is expected to be in the range 0-127 hence the division by 127 * 127 to
* get back into range. */
uint8 effect_vol_setting = _effect_vol.load(std::memory_order_relaxed);
uint8 effect_vol = (effect_vol_setting *
effect_vol_setting *
effect_vol_setting) / (127 * 127);
/* Mix each channel */
uint8 active = _active_channels.load(std::memory_order_acquire);
for (uint8 idx : SetBitIterator(active)) {
MixerChannel *mc = &_channels[idx];
if (mc->is16bit) {
mix_int16(mc, (int16*)buffer, samples, effect_vol);
} else {
mix_int8_to_int16(mc, (int16*)buffer, samples, effect_vol);
}
if (mc->samples_left == 0) MxCloseChannel(idx);
}
}
MixerChannel *MxAllocateChannel()
{
uint8 currently_active = _active_channels.load(std::memory_order_acquire);
uint8 available = ~currently_active;
if (available == 0) return nullptr;
uint8 channel_index = FindFirstBit(available);
MixerChannel *mc = &_channels[channel_index];
free(mc->memory);
mc->memory = nullptr;
return mc;
}
void MxSetChannelRawSrc(MixerChannel *mc, int8 *mem, size_t size, uint rate, bool is16bit)
{
mc->memory = mem;
mc->frac_pos = 0;
mc->pos = 0;
mc->frac_speed = (rate << 16) / _play_rate;
if (is16bit) size /= 2;
/* adjust the magnitude to prevent overflow */
while (size >= _max_size) {
size >>= 1;
rate = (rate >> 1) + 1;
}
mc->samples_left = (uint)size * _play_rate / rate;
mc->is16bit = is16bit;
}
/**
* Set volume and pan parameters for a sound.
* @param mc MixerChannel to set
* @param volume Volume level for sound, range is 0..16384
* @param pan Pan position for sound, range is 0..1
*/
void MxSetChannelVolume(MixerChannel *mc, uint volume, float pan)
{
/* Use sinusoidal pan to maintain overall sound power level regardless
* of position. */
mc->volume_left = (uint)(sin((1.0 - pan) * M_PI / 2.0) * volume);
mc->volume_right = (uint)(sin(pan * M_PI / 2.0) * volume);
}
void MxActivateChannel(MixerChannel *mc)
{
uint8 channel_index = mc - _channels;
_active_channels.fetch_or((1 << channel_index), std::memory_order_release);
}
/**
* Set source of PCM music
* @param music_callback Function that will be called to fill sample buffers with music data.
* @return Sample rate of mixer, which the buffers supplied to the callback must be rendered at.
*/
uint32 MxSetMusicSource(MxStreamCallback music_callback)
{
std::lock_guard<std::mutex> lock{ _music_stream_mutex };
_music_stream = music_callback;
return _play_rate;
}
bool MxInitialize(uint rate)
{
std::lock_guard<std::mutex> lock{ _music_stream_mutex };
_play_rate = rate;
_max_size = UINT_MAX / _play_rate;
_music_stream = nullptr; /* rate may have changed, any music source is now invalid */
return true;
}
void SetEffectVolume(uint8 volume)
{
_effect_vol.store(volume, std::memory_order_relaxed);
}
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